Re: [asterisk-users] asterisk-users Confbridge
Sorry about the trouble. Unsubscribed that user from the mailing lists. Matthew Fredrickson On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote: > > I'm online on this site! > So contact me in my profile: > here > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken
On Mon, Jul 13, 2020 at 2:34 PM Saint Michael wrote: >> >> There is a big confusion here about Stir Shaken. It is NOT a provider issue. >> Un fact, all providers are whasing their hands and modifying their swihtches >> to pass-through the Signature. They cannot sign the call because then the >> become the responsible party for the call before the FCC, and liable for any >> illegal call. Every owner of a PBX that sends calls to the network, except >> if you use a trunk for the likes of Vonage, needs to sign their calls. So if >> you send calls with any kind of dialer and use DIDs, real or "borrowed", you >> need to get the signature service urgently or your business will stop >> terminating calls. You cannot self-sign, you cannot get around it, the calls >> will either go to straight to voicemail or fail. Even worse, the carries wil >> play a fake voicemail and charge you a fee, something that some already a >> are doing when they detect robocallig. > > Don't even think about Transnexus, because they use 302 Redirect with a > header, and no version of Asterisk supports it. I am the only game in the > world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is > literally true. If you need to sign your calls to get through, with Asterisk, > you need to connect to my service. I am an approved Service Provider from the > FCC. If you keep thinking this is not happening, it is, and your business > will disappear overnight. > The issue is that Vicidial, for example, does not provide res_odbc and > func_odbc, so you need to solve that first with Vicidial. Then you can apply > the code I provided earlier and your calls with have a legal, binding > signature. The carriers verify each signature and discard the ones that fail > the cryptography test. Sounds like you're trying to sell/direct people towards a service that you've created. Feel free to do so on the -biz list but the -users list isn't the right place for that sort of thing. Matthew Fredirckson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stir Shaken is upon us
On Sun, Jul 12, 2020 at 5:18 PM Joshua C. Colp wrote: > > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote: >> >> Asterisk 18 will have support based on this asterisk update Matt F did for >> CommCon's sponsor slots >> >> https://youtu.be/eas1csaX-wc >> > > As well support will go into Asterisk 16 and 17 as well. It's just been under > active development so we've been waiting for that to finish before bringing > it back into those versions. > Thanks for clarifying that Josh. I only had 5 min on the CommCon presentation so I focused more on the Asterisk 18 side of things rather than clarifying a lot of that :-) Matthew Fredrickson > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Usage Survey
Hey All, For those of you that do not know me, my name is Matthew Fredrickson and I’m the project lead for the Asterisk project. First off, I wanted to thank all of you that contribute in various ways to the project – whether it be at a developmental level, answering questions on forums and mailing lists, contributing documentation, or just generally advocating for it within your sphere of influence. It takes so many people’s efforts to make the project what it is and to sustain such a large and vibrant user and developer community. We created a general survey inquiring how people utilize Asterisk. It should only take about 10-15 minutes, but would help us understand better how our users are utilizing Asterisk and help us to understand if there are important areas of Asterisk that we underemphasize from a development perspective. If you don’t mind filling it out, it would be greatly appreciated. Thanks *so* much again for your time, and best wishes to each of you in your efforts. https://goo.gl/forms/xL1VUHRsf95saly13 Matthew Fredrickson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tel URI
On Thu, Jan 31, 2019, 9:24 AM Jean-Denis Girard Hi list, > > Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a > system that uses exclusively tel: uri on inbound and outbound calls. I > could not find documentation or sample config about tel:uri. Is this > doable? If not possible with PJSIP, is chan_sip a better option? Any > pointer would be greatly appreciated. > Right now, chan_pjsip does not properly handle tel: URIs. If you need them you might need to use chan_sip. Matthew Fredrickson > > Thanks, > -- > Jean-Denis Girard > > SysNux Systèmes Linux en Polynésie française > https://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.797.527 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stop
Unsubscribe info is in the footer of the message Best wishes, Matthew Fredrickson On Mon, Oct 1, 2018, 6:22 AM Karen York wrote: > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez wrote: > > On 9/26/2018 4:46 AM, Olivier wrote: > > > Hello, > > > > This morning, I asked myself if WebRTC could be a viable alternative > > to softphone deployment. > > > > For me, main issue with Softphones is the amount of work needed for > > installation and configuration. > > Also, Softphones must be carefully choosen if Deskphone-like quality > > is expected. > > > > Now that WebRTC becomes ubiquitous, it might make sense to trade > > Softphone features (call history, BLF, ...) for WebRTC deployment > > simplicity. > > > > What do you think of this ? > > What kind of experience did you met with such WebRTC deployments ? > > What about classic telephony features (CallTransfer) ? > > Have you tried Cyber Maga Phone 2K ? > > > > If you can get it to work WebRTC is a good option. The problem is > that any changes in your network may disrupt it and even trying to > replicate your installation is difficult. I have it working fine on my > website so customers can call us directly from our web page but I never > could get Cyber Mega Phone 2K to work on the same server. We used JSSIP > to create the webrtc phone on our website. We just updated the documentation for how to get CMP2K working on the wiki [1]. We'd love some feedback if you still have issues getting it setup so that we can improve the docs. [1] https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone Best wishes, Matthew Fredrickson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI timeout option
Hey,
I would suggest starting a new thread with this question instead of
inserting this into another existing thread like this.
Matthew Fredrickson
On Tue, Sep 18, 2018, 11:16 AM modou lo wrote:
> Please can i ask you i want to know which code can help me to provide the
> taxation of voip/toip services in asterisk
>
> Le mar. 18 sept. 2018 à 01:36, Patrick Wakano a
> écrit :
>
>> Thanks everyone for the answers!
>> I did explored some options at the PHP level and probably will do
>> something in this direction, but in fact what I was really looking was
>> something in the Asterisk side, not in the script side.
>> Because in my opinion regardless of the language or AGI type, Asterisk
>> itself should be able to timeout a long running script and return to the
>> dialplan. However looks like there is nothing of this sort.
>>
>> Kind regards,
>> Patrick Wakano
>>
>> On Sat, 15 Sep 2018 at 03:56, Eric Wieling wrote:
>>
>>> I don't know AGIspeedy, but I have some PHP scripts where I set a
>>> connect timeout using streams.
>>>
>>> Example using https, but should be easily adaptable to non-s http.:
>>>
>>> $pbxsh_bin = @file_get_contents("https://blah.blah.blah;, FALSE,
>>> @stream_context_create(array('https' => array('timeout' => 5,
>>> "verify_peer"=>false, "verify_peer_name"=>false;
>>>
>>> On 09/14/2018 01:40 PM, Carlos Chavez wrote:
>>> > On 9/13/2018 8:04 PM, Patrick Wakano wrote:
>>> >
>>> >> Hello list,
>>> >> Hope you all doing well!
>>> >>
>>> >> Recently, I had an issue with a FastAGI PHP script, which under some
>>> >> specific situation would run into an infinity loop, consuming all CPU
>>> >> resources. This also was preventing Asterisk to terminated the call
>>> >> properly because it was waiting for the AGI to return... The
>>> >> application uses AGIspeedy to process the AGI calls, not sure if this
>>> >> can be affecting this situation somehow
>>> >> Due to this problem I started looking for some option to timeout the
>>> >> AGI call and return to the dialplan after XYZ seconds and so this
>>> >> would protect Asterisk preventing the dialplan to get stuck due to
>>> >> some external script problem that is actually outside of Asterisk
>>> >> control. Does Asterisk provide some control of this sort? I searched
>>> >> around and could not find any.
>>> >> Any idea is appreciated!
>>> >>
>>> >> Kind regards
>>> >> Patrick Wakano
>>> >>
>>> >
>>> > I think this is what you may be looking for:
>>> >
>>> > http://php.net/manual/en/function.set-time-limit.php
>>> >
>>>
>>> --
>>> http://help.nyigc.net/
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Astricon is coming up October 9-11! Signup is available at:
>>> https://www.asterisk.org/community/astricon-user-conference
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Astricon is coming up October 9-11! Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11! Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 16 AMI changes
Usually yes. You'll need to read the UPGRADE.txt and CHANGES files to get a good idea of the specific changes though. Best wishes, Matthew Fredrickson On Thu, Sep 6, 2018, 7:44 PM Telium Support Group wrote: > Does anyone know if Asterisk 16 includes changes to the AMI? (syntax / > commands / etc) > > > > I see a release candidate is forthcoming. Just curious > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More testing
:-) Sorry to disappoint. On Wed, May 23, 2018, 10:21 AM John Kiniston <johnkinis...@gmail.com> wrote: > I got excited when I saw 8 new messages on the Asterisk list-serve this > morning, What discussions must be happening I thought! > > You are a tease sir. > > On Tue, May 22, 2018 at 7:58 PM, Matt Fredrickson <cres...@digium.com> > wrote: > >> More testing. Test test test. :-) >> >> -- >> Matthew Fredrickson >> Digium, Inc. | Engineering Manager >> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > A human being should be able to change a diaper, plan an invasion, butcher > a hog, conn a ship, design a building, write a sonnet, balance accounts, > build a wall, set a bone, comfort the dying, take orders, give orders, > cooperate, act alone, solve equations, analyze a new problem, pitch manure, > program a computer, cook a tasty meal, fight efficiently, die gallantly. > Specialization is for insects. > ---Heinlein > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Testing for real from a non-digium email
Here we go! Matt -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / IAX / Problems with incoming calls
As I recall, there was an IAX2 protocol addition for newer versions of Asterisk a while ago due to a security issue - which can potentially trigger IAX2 interop issues if your config file for chan_iax2 is not setup properly. You can read more about it here: http://downloads.asterisk.org/pub/security/IAX2-security.pdf With regards to the CTOKEN addition. Hope that helps. Matthew Fredrickson Digium, Inc. On 3/8/13 8:38 AM, Thorsten Göllner wrote: Hi, I have upgraded vom Atserisk 1.6.1.20 to 11.2.1. Most things went fine. But 1 thing will not work: IAX. I used the same configuration but Asterisk will not answer the incoming IAX-Call. When enabling iax debugging I can see the following: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:VERSION : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:39] VERBOSE[3219] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:VERSION : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLED NUMBER : 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CODEC_PREFS : (alaw|ulaw|gsm|speex16|g729|g723) [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NUMBER : 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING PRESNTN : 3 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TYPEOFN : 0 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING TRANSIT : 1 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLING NAME: 0049... [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:LANGUAGE: en [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:USERNAME: 02070992875 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:FORMAT : 8 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CAPABILITY : 65535 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:ADSICPE : 2 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:DATE TIME : 2013-03-07 16:14:38 [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: CTOKEN [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:Timestamp: 4ms SCall: 1 DCall: 05992 [77.240.54.23:4572] [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c:CALLTOKEN : 51 bytes [Mar 7 17:14:41] VERBOSE[3220] chan_iax2.c: [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:43] VERBOSE[3221] chan_iax2.c: [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:Timestamp: 04006ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c:CAUSE CODE : 0 [Mar 7 17:14:45] VERBOSE[3222] chan_iax2.c: [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c: Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW [Mar 7 17:14:51] VERBOSE[3223] chan_iax2.c:Timestamp: 4ms SCall: 05992 DCall: 0 [77.240.54.23:4572] [Mar
Re: [asterisk-users] Can't detect remote answer
Hey, Just quickly glanced over your data... one problem you have is that you're passing the 'r' flag in your Dial() statement in extensions.conf. That would definitely cause you to have never ending ringback from the analog line (since answer supervision is often not present). You might try removing that and retry your outbound call test. Hope that helps a bit. Matthew Fredrickson Digium, Inc. On 2/11/13 10:54 AM, Kevin Wright wrote: I forgot to add, cat /proc/dahdi/* yields: Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS (In use) (EC: MG2 - INACTIVE) 2 WCTDM/4/1 FXOKS (EC: MG2 - INACTIVE) 3 WCTDM/4/2 Reserved 4 WCTDM/4/3 Reserved I'm not sure if that (in use) is correct when I'm not actively in a call. This is a very sensitive setup, as a home installation it absolutely *must* pass the gruelling wife test, so I'm keen to see it up and running properly :) On 11 February 2013 16:50, Kevin Wright kev.lee.wri...@gmail.com mailto:kev.lee.wri...@gmail.com wrote: I'm attempting to place an outgoing call over POTS/DAHDI, it dials without problem but the remote answer isn't tried. So far I've attempted: * Searching on google * Enabling full and verbose logging (including the debug option of the DAHDI module) - showing NO event at the time I answer on the remote phone a.k.a my mobile * Using another phone on the same line - it works * Receiving a call on that line - no problem * Logging DTMF - it shows digits dialled on my mobile, after I've answered, even whilst it seems to still be ringing locally * Looking on the wiki * Asking on IRC So far, I've found nothing that helps. A sample log output is here: http://pastebin.com/cprZSy9i And my chan_dahdi.conf: http://pastebin.com/L7mBJ66Y And dahdi system.conf: http://pastebin.com/6UQPVC9x also in modprobe.d/dahdi.conf: http://pastebin.com/5ZqtcZdj *any* advice/suggestions at this point would be very much appreciated! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi timing source multiple cards
You must make sure that for each card, the timing parameter does not exceed the number of spans on the card (unless you're using a timing cable between cards). So you probably don't want to have anything above a 4 for the timing parameter... I see below that you have 5-12 listed in the timing parameter for the spans on the other cards. You probably want something more like this: span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs span=5,1,0,esf,b8zs span=6,2,0,esf,b8zs span=7,3,0,esf,b8zs span=8,4,0,esf,b8zs span=9,1,0,esf,b8zs span=10,2,0,esf,b8zs span=11,3,0,esf,b8zs span=12,4,0,esf,b8zs Hope that helps. Matthew Fredrickson Digium, Inc. On 12/20/12 10:42 PM, Dave George wrote: I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72 span=4,4,0,esf,b8zs bchan=73-96 span=5,5,0,esf,b8zs bchan=97-120 span=6,6,0,esf,b8zs bchan=121-144 span=7,7,0,esf,b8zs bchan=145-168 span=8,8,0,esf,b8zs bchan=169-192 span=9,9,0,esf,b8zs bchan=193-216 span=10,10,0,esf,b8zs bchan=217-240 span=11,11,0,esf,b8zs bchan=241-264 span=12,12,0,esf,b8zs bchan=265-288 dahdi_scan : [1] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=1 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=25 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [3] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 3 name=TE4/0/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=49 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [4] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 4 name=TE4/0/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=Board ID Switch 0 basechan=73 totchans=24 irq=225 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [5] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 1 name=TE4/1/1 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=97 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [6] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 2 name=TE4/1/2 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=121 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [7] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 3 name=TE4/1/3 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=145 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [8] active=yes alarms=OK description=T4XXP (PCI) Card 1 Span 4 name=TE4/1/4 manufacturer=Digium devicetype=Wildcard TE410P (3rd Gen) (VPMOCT128) location=PCI Bus 10 Slot 03 basechan=169 totchans=24 irq=233 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [9] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 1 name=TE4/2/1 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=193 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [10] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span 2 name=TE4/2/2 manufacturer=Digium devicetype=Wildcard TE410P (4th Gen) (VPMOCT128) location=PCI Bus 10 Slot 04 basechan=217 totchans=24 irq=50 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [11] active=yes alarms=OK description=T4XXP (PCI) Card 2 Span
Re: [asterisk-users] Digium TE205P leds flash red on startup
On 12/15/11 12:47 PM, Vieri wrote: ZT_SPANCONFIG failed on span 1: No such device or address (6) The fact that there's nothing in /proc/zaptel/ makes me think that the zaptel kernel module isn't working. Is the 1205 card compatible with zaptel 1.4.12.1? (I can't migrate to DAHDI on this system - at least not yet) Thanks, Vieri -- That's the reason why it's not working. Unfortunately, the newer versions of those cards require DAHDI in order to operate. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On 11/18/10 7:40 AM, Matt wrote: On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com wrote: On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew, So 1.6.0 or newer, with a Digium card should talk T1/SS7 no problem? That's great news! I have a Nortel DMS100 that we have configured for DS1/SS7 and we were trying to figure out how to connect an Asterisk PBX to it. That is correct. Feel free to ask me any questions if you have any issues come up along the way. The sample chan_dahdi.conf has a section with an example of an SS7 setup in it, for reference on configuration. Matthew Fredrickson Hardware/Software Engineer Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On 11/18/10 10:07 AM, Matthew Fredrickson wrote: On 11/18/10 7:40 AM, Matt wrote: On Wed, Nov 17, 2010 at 6:17 PM, Matthew Fredricksoncres...@digium.com wrote: On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew, So 1.6.0 or newer, with a Digium card should talk T1/SS7 no problem? That's great news! I have a Nortel DMS100 that we have configured for DS1/SS7 and we were trying to figure out how to connect an Asterisk PBX to it. That is correct. Feel free to ask me any questions if you have any issues come up along the way. The sample chan_dahdi.conf has a section with an example of an SS7 setup in it, for reference on configuration. Oh yeah, and also, there's an asterisk-ss7 mailing list at lists.digium.com where SS7 related discussion and questions usually take place. Matthew Fredrickson Hardware/Software Engineer Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
On 11/17/10 2:44 PM, Cary Fitch wrote: In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Asterisk (as of version 1.6.0 or greater) has native support for SS7 with DAHDI interface cards in chan_dahdi. I obviously have used it with quite a few Digium cards that have worked well. Matthew Fredrickson Hardware/Software Engineer Digium, Inc. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Sent: Wednesday, November 17, 2010 2:31 PM To: asterisk-users Subject: [asterisk-users] GSM and SS7 Questions I have two questions for the group. #1 - I'm looking to use some GSM SIM cards with my Asterisk PBX. Can anyone recommend a gateway? I need about 10-15 SIM slots. #2 - I'm also looking to connect Asterisk to an SS7 signaled DS1 (24 channels) for inbound and outbound voice calls. Can anyone offer any suggestions for cards to use there? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED
This is not actually a problem... it's a side affect of how older versions of libpri handled PTMP links. Basically, after 3-5 minutes, the other side is probably trying to drop layers 1 and 2 due to no calls being active. For the most part, unless you see any issues, you should just ignore the message. This is just libpri re-establishing layer when the other side tries to drop it, due to its desire to have the perception of a persistent layer 2 (in older versions). In newer libpri (1.4 branch) it allows layer 2 to drop and stay dropped until it is needed by layer 3. Matthew Fredrickson Digium, Inc. Darshaka Pathirana wrote: Hi everyone. We have a problem here... Hope somebody can give us some hints. We have a HP ProLiant DL180 G6 Server with a Debian/Lenny sytem. Asterisk 1.4.21.2 (1.4.21.2~dfsg-3+lenny1) with zaptel (1.4.11) and libpri (1.4.3) is installed. There is a QuadBRI-Card installed: # lspci -vv -s 06:04.0 06:04.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) Subsystem: Cologne Chip Designs GmbH Device b752 Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- INTx- Interrupt: pin A routed to IRQ 30 Region 0: I/O ports at cc00 [size=8] Region 1: Memory at fb6ff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- zttest gives me an average of 99.992% and zttool shows no alarms. But every about 3,5 minutes we get this (with debug span 1 enababled): 1 -- Timeout occured, restarting PRI 1 q921.c:859 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED 1 Sending Set Asynchronous Balanced Mode Extended 1 q921.c:534 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Apr 10 12:16:05] WARNING[28541]: chan_zap.c:2498 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway! 1 Sending Set Asynchronous Balanced Mode Extended 1 -- Got UA from network peer Link up. 1 -- Restarting T203 counter == Primary D-Channel on span 1 up % cat /etc/zaptel.con # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: ztqoz/1/1 quadBRI PCI ISDN Card 1 Span 1 [TE] (cardID 0) (MASTER) span=1,1,3,ccs,ami # termtype: te bchan=1-2 dchan=3 # Span 2: ztqoz/1/2 quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 0) span=2,2,0,ccs,ami # termtype: te bchan=4-5 dchan=6 # Span 3: ztqoz/1/3 quadBRI PCI ISDN Card 1 Span 3 [TE] (cardID 0) span=3,3,0,ccs,ami # termtype: te bchan=7-8 dchan=9 # Span 4: ztqoz/1/4 quadBRI PCI ISDN Card 1 Span 4 [TE] (cardID 0) span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 # Global data loadzone= at defaultzone = at % cat /etc/asterisk/zapata.conf [channels] language=de switchtype=euroisdn pridialplan=unknown prilocaldialplan=dynamic priindication=passthrough context=incoming immediate=no usecallingpres=yes usecallerid=yes group=1 nationalprefix=00 internationalprefix=000 signalling=bri_cpe echocancel=Yes overlapdial=Yes ; group=2 ; signalling=bri_cpe ; context=incoming ; channel = 10-11 ; channel = 1-2 ; channel = 4-5 ; channel = 7-8 ; channel = 10-11 (Only one span is connected to ISDN right now.) qozap is loaded and ztcfg -v gives me: Zaptel Version: 1.4.11 Echo Canceller: MG2 Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) 12 channels to configure. Any idea what this could mean and how this could be fixed? Any help would be helpful. Thx. Greetings, - Darsha -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone coming to Paris next week for AstriEurope?
Randy R wrote: Several regulars from the VUC will be there, some of us are arriving Tuesday night. Anyone else considering the trip? Post here or contact me off list so we can meet. /r I'll be there... For those that don't know me, I work a lot on chan_dahdi/libss7/libpri/DAHDI. I'm not sure what my schedule is going to be like there, but I'd love to hear about any meetups that may happen if I can fit it in. Matthew Fredrickson Digium, Inc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
Martin wrote: pri debug span 1 should show you the ISDN messages for 2BCT if there are any Also someone should have told you that the 2BCT code is by default not compiling and you could enable it by editing chan_dahdi.c and adding #define PRI_2BCT Also since this flag is not present anywhere else in the code grep PRI_2BCT * -r channels/chan_dahdi.c:#ifdef PRI_2BCT channels/chan_dahdi.c:#ifdef PRI_2BCT it might actually only work in the version of Asterisk it was introduced for ... This flag is defined inside of libpri.h, which is included by chan_dahdi.c, which is why you do not see it inside chan_dahdi.c. 2BCT will automatically compile by default if the version of libpri you have support 2BCT. If you have any version of libpri newer than a year or two ago, it supports all the currently supported switchtypes. In fact, the earliest version of 2BCT supported was done probably 3 or 4 years ago (RLT for DMS switches), so even very old versions of libpri will support compilation of that code. Matthew Fredricikson Digium, Inc. Martin On Wed, Apr 15, 2009 at 8:24 AM, Ron Joffe rjo...@sienatech.com wrote: On Tuesday 14 April 2009 18:41, Jared Smith wrote: Some time after the second leg of the call has answered, Asterisk will send a facility message to the CO switch saying Hey, mind bridging these two calls on your end, so I can free up the channels on my end? If the switch says OK, you'll see the calls disappear from Asterisk (and the people on the calls won't know the difference). Otherwise, the calls will continue to be bridged by Asterisk. Jared, Is there a debug mode where I can find these specific messages? Thanks, Ron -- Ron Joffe Siena Tech, Inc. 3319 Willow Glen Drive Oak Hill, VA 20171 (919) 928-0404 ___ -- 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] 2B Channel Transfer on XO-based T1
Jared Smith wrote: On Wed, 2009-04-15 at 09:58 -0500, Kevin P. Fleming wrote: It's not enabled by default because when it is used the Asterisk server loses control of the call and the CDR becomes incomplete. Not everyone wants that behavior. But since many people *would* like that behavior, wouldn't it make more sense to enable this via an option in chan_dahdi.conf? Maybe enable2bct=yes? (It's not like you don't already have to set facilityenable=yes and transfer=yes to get it anyway, and I doubt there are many people who want facilityenable=yes and transfer=yes but not 2bct... But for those few, I guess we can add yet another option.) It seems silly to have to recompile just to get this functionality. It *is* compiled in by default and it actually *is* configurable. I've said this a few times in the archives, but just so that everyone knows, in order for it to work, 'transfer=yes' must be set in chan_dahdi.conf on each of the channels you would like to enable it on. To disable it for a channel or group of channels, set 'transfer=no' above that group. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2B Channel Transfer on XO-based T1
Don Kelly wrote: Someone referred to a facility message when the TBCT call is torn down. There are actually two messages--when the PSTN switch takes back the calls and completes the transfer, it sends a facility message including a unique ID. Then, when one of the parties disconnects, the switch sends another facility message with the same unique ID. This would provide information to complete the CDR record. Now that there seems to be some interest in TBCT, is someone interested in handling these two facility messages to update the CDR? Unfortunately, when I implemented this code I did not add support for this feature since it would probably have required some core changes to do so. So right now, we simply ignore that message and go about our merry way. Matthew Fredrickson Digium, Inc. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Wednesday, April 15, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 2B Channel Transfer on XO-based T1 Philipp Kempgen wrote: Could somebody shed some light on why PRI_2BCT is not enabled by default? Is it an experimental feature? I'd like to compile stuff without patching defines. :-) It's not enabled by default because when it is used the Asterisk server loses control of the call and the CDR becomes incomplete. Not everyone wants that behavior. ___ -- 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] 2BCT last mile... Hopefully
Max Metral wrote: Ok, so I’ve made progress on 2BCT (2 B-Channel Transfer). I’m assuming that the debug info below shows that XO doesn’t have 2BCT enabled on my line, but if anybody can confirm that’ll let me be way more indignant. J It would appear that the switch doesn't like what you're sending it. That means either your switch at the other end is not configured with this feature enabled or that your switchtype is set incorrectly for the actual switch type that the other end is expecting. From the message you are sending to the other side, it would appear that you are configured for either 5ESS or national switch type. Another possiblity (although low in probability) is that it doesn't like being sent the transfer message so soon, since it would appear that we have not yet received the CONNECT-ACK from the other switch by the time we send the transfer request. You could try inserting a Wait(5) after you Answer() the call in your dialplan before Dial()'ing back out to verify that the call is completely setup. Make sure you try explicitly Answer()'ing the call first in your dialplan before Wait()'ing or Dial()'ing back out, at least until you figure out what the problem is. Matthew Fredrickson Digium, Inc. -- Native bridging DAHDI/1-1 and DAHDI/3-1 Protocol Discriminator: Q.931 (8) len=28 Call Ref: len= 2 (reference 801/0x321) (Terminator) Message type: FACILITY (98) [1c 15 91 a1 12 02 01 23 06 07 2a 86 48 ce 15 00 08 30 04 02 02 01 93] Facility (len=23, codeset=0) [ 0x91, 0xA1, 0x12, 0x02, 0x01, '#', 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x04, 0x02, 0x02, 0x01, 0x93 ] PROTOCOL 11I A1 0012 (CONTEXT SPECIFIC [1]) 02 0001 23 (INTEGER: 35) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0004 (SEQUENCE) 02 0002 01 93 (INTEGER: 403) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 801/0x321) (Originator) Message type: CONNECT ACKNOWLEDGE (15) q931.c:3705 q931_receive: call 801 on channel 1 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=16 Call Ref: len= 2 (reference 801/0x321) (Originator) Message type: FACILITY (98) [1c 09 91 a3 06 02 01 23 02 01 00] Facility (len=11, codeset=0) [ 0x91, 0xA3, 0x06, 0x02, 0x01, '#', 0x02, 0x01, 0x00 ] PROTOCOL 11I A3 0006 (CONTEXT SPECIFIC [3]) 02 0001 23 (INTEGER: 35) 02 0001 00 (INTEGER: 0) -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE return error component Unable to handle return result on switchtype 1! ___ -- 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] [astersik-users] ss7 consultancy $1000 USD
Apu Islam wrote: I am looking for someone who will implement an ss7 to sip media server. email me personally for more details. I expect a professional attitude and preferably someone I can communicate in English. There will be phone conversations and IM communications. If you do not have experience implementing this, please do not reply. Bounty is $1000 USD, will be paid cash with signed contract. Thanks. This is a pretty simple thing to do using Asterisk with its native SS7 stack (libss7). You might even be able to do it yourself. In any case, a lot of the people that have experience doing things like this are actually on the Asterisk-SS7 mailing list (you can subscribe to it at lists.digium.com). Please let me know if you have any issues with using and/or configuring libss7, since I very much want it to be easy for people to use and configure. (I wrote it) :-) -- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune and the bug
bilal ghayyad wrote: Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Current version of fxotune (in current 1.4 Zaptel and DAHDI) does not have any outstanding bugs. From a quick glance over the wiki page, it looks like it has some interesting information, but a lot of it is out of date. My guess is the bug you're referring to is the one that says it has problems with dialtone detection or something of that nature. The most current version of fxotune is pretty much immune to dialtone or other background noise due to the newer way it does signal measurement (using frequency analysis instead of frequency agnostic power calculation), so you shouldn't see any problems with this. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune and the bug
bilal ghayyad wrote: Dear Mathew; Kindly find the link of the batch tha fixed the bug: http://bugs.digium.com/view.php?id=7136 It is written that last update was in 2008-06-07 11:36, so for that I do not know if my asterisk and zaptel versions include this fix or not? Because I installed them before this date. How can I know starting from which version this patch has been included? That particular patch is old and out of date and does not have the latest fixes that include the background noise and tone immunity code. If your problem is that you simply don't want to update Zaptel though, you can build use the fxotune utility from the latest version of Zaptel and just don't run make install so you don't overwrite your existing Zaptel. Matthew Fredrickson Digium, Inc. Any advise. Regards Bilal --- On Thu, 4/2/09, Matthew Fredrickson cres...@digium.com wrote: From: Matthew Fredrickson cres...@digium.com Subject: Re: [asterisk-users] fxotune and the bug To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thursday, April 2, 2009, 12:17 PM bilal ghayyad wrote: Hi All; I got to know (reading on the wiki) that fxotune was have a bug, and it has been fixed. But I do not know if my current asterisk version contain the fixed one or not? How can I know? My current asterisk version is 1.4.22 Current version of fxotune (in current 1.4 Zaptel and DAHDI) does not have any outstanding bugs. From a quick glance over the wiki page, it looks like it has some interesting information, but a lot of it is out of date. My guess is the bug you're referring to is the one that says it has problems with dialtone detection or something of that nature. The most current version of fxotune is pretty much immune to dialtone or other background noise due to the newer way it does signal measurement (using frequency analysis instead of frequency agnostic power calculation), so you shouldn't see any problems with this. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_ss7 with ringing, but no voice stream.
Cary Fitch wrote:
SS7 doesn’t send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7 code, there could be some error in setting
up the call, but normally the IMT trunks are two way. (Of course they are 4
wire circuits so are two one way paths, but they are matched pairs so,
for practical purposes they would be 1 entity for call set up purposes.)
Actually, the implementations of SS7 support in Asterisk (libss7, and
also the out of tree chan_ss7) include support for signaling and bearer
channels, which is why he's mentioning voice support.
Right now, both implementations function basically like the ISDN code
works - i.e. you have to terminate signaling and bearer channels on the
same box.
Matthew Fredrickson (the libss7 guy :-) )
Digium, Inc.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of lizhong zhu
Sent: Friday, March 20, 2009 2:05 AM
To: asterisk-ss7
Subject: [asterisk-users] chan_ss7 with ringing, but no voice stream.
hello, all of users:
sorry, resend it again for clarifying the message. I have implemented
cha_ss7 in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think the ss7 does not send the voice steam to the destination.
in chan_ss7, i added:
===
static struct ss7_chan *cic_hunt_even_mru(struct linkset*
linkset) {
struct ss7_chan *cur, *prev, *best, *best_prev;
best = NULL;
best_prev = NULL;
for(cur = linkset-idle_list, prev = NULL; cur !=
NULL; prev = cur, cur = cur-next_idle) {
/* Don't select lines that are resetting or
blocked. */
if(!cur-reset_done || (cur-blocked
(BL_LH|BL_RM|BL_RH|BL_UNEQUIPPED|BL_LINKDOWN))) {
continue;
}
/* if((cur-cic % 2) == 0) { */
/*change to this*/
if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
/* Choose the first idle even circuit,
if any. */
/*end of change*/
best = cur;
best_prev = prev;
break;
} else if(best == NULL) {
/* Remember the first odd circuit, in
case no even circuits are
available. */
best = cur;
best_prev = prev;
}
}
cic_hunt_even_mru if(((cur-cic % 2) ==
0)0==strcasecmp(cur-link-name,linkname))
{
my environment is:
asterisk-1.4.20
chan_ss7-1.0.91
Openvox D410P
===
anyone has an idea for the problem?
please give me some hints!
thanks!
james.zhu
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Re: [asterisk-users] Asterisk 1.6, B410P and TE/PtmP mode. Could you get it running ?
Olivier wrote: Hi, My setup is: IPPhone1 --- Asterisk1 with B410P Patton 4638 --- Asterisk2 --- IPPhone2 I want to evaluate Asterisk1 in TE/PtmP mode. So, Patton box is configured in NT/PtmP (with 3 BRI links between both systems). Anyway, asterisk -rx pri show spans keeps replying : PRI span 1/0: Provisioned, Down, Active PRI span 2/0: Provisioned, Down, Active PRI span 3/0: Provisioned, Down, Active I came accross this : http://bugs.digium.com/view.php?id=14031 Unfortunately, enclosed patch doesn't apply to asterisk 1.6.0.6. My question is: could anyone make asterisk work in TE/PtmP with a B410P ? Hey Olivier, I wrote most of this code, and would be very interested in asking you a little more about this issue. Can you IM me? On MSN, I am creslin...@hotmail.com AOL: MatthewFredricks jabber: cres...@digium.com Thanks, Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi wcb4xxp and fax
Olivier wrote: 2009/2/27 Matthew Fredrickson cres...@digium.com mailto:cres...@digium.com I have a couple of suggestions: Make sure that your timing configuration is correct in /etc/dahdi/system.conf (that it has a valid timing source). Also, you probably will probably want to use the half_full buffer policy, and set the number of buffers used to something reasonable, like 8, to ensure you don't have any transmit buffer underruns on the B410P. You shouldn't need more than that, since you're not trying to deal with clock slips or timing drift in this configuration. Which buffer is it referred to here ? In which file should such settings happen ? It is the bufferpolicy option in /etc/asterisk/chan_dahdi.conf. If you will get in contact with me via IM, I'd like to see if we can figure out what's going on here. Thanks, Matthew Fredrickson Digium, Inc. AIM: MatthewFredricks MSN: creslin...@hotmail.conf Jabber: creslin2digium.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] Sending faxes with T.38 problem. Asterisk - 1.6.0.6
Santiago Gimeno wrote: Hello, Thanks everybody for the answers. Could be. Would you post the Cisco config relevant to this? dial-peer voice 5 voip description ** ** preference 1 destination-pattern 1… voice-class codec 1 session protocol sipv2 session target ipv4:1.1.1.1 session transport udp dtmf-relay rtp-nte fax-relay ecm disable I think, that at least if you're using T.38, you may want to try enabling ECM. ECM can cause significant problems in a high-packet loss, non-T.38 environment, but I would think that in a T.38 environment, if you can keep ECM enabled, that would be a good thing. Matthew Fredrickson Digium, Inc. fax nsf 00 fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw no vad And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. I have removed the T38CALL variable and it looks better but it still doesn't work. Now asterisk sends an initial INVITE with audio media in the SDP. The CISCO accepts this call after contacting the fax-machine. Then the CISCO sends a re-INVITE with the T.38 SDP. Asterisk accepts this re-INVITE. But finally the fax transmission fails and the asterisk verbose trace is: *CLI -- Attempting call on SIP/080913216...@outbound-calls for 22...@fax-out:1 (Retry 1) == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Channel SIP/outbound-calls-0822aae8 was answered. == Starting SIP/outbound-calls-0822aae8 at fax-out,2,1 failed so falling back to exten 's' -- Executing [...@fax-out:1] Set(SIP/outbound-calls-0822aae8, FAXFILE=/root/santi/fax/prueba.tif) in new stack -- Executing [...@fax-out:2] SIPDtmfMode(SIP/outbound-calls-0822aae8, inband) in new stack -- Executing [...@fax-out:3] SendFAX(SIP/outbound-calls-0822aae8, /root/santi/fax/prueba.tif) in new stack [Mar 10 17:15:28] WARNING[17125]: app_fax.c:176 phase_e_handler: Error transmitting fax. result=11: Far end cannot receive at the resolution of the image. [Mar 10 17:15:28] WARNING[17125]: app_fax.c:621 transmit: Transmission error == Spawn extension (fax-out, s, 3) exited non-zero on 'SIP/outbound-calls-0822aae8' Any ideas? Thanks. Best regards, Santi On Tue, Mar 10, 2009 at 4:26 PM, Joshua Colp jc...@digium.com mailto:jc...@digium.com wrote: - Santiago Gimeno santiago.gim...@gmail.com mailto:santiago.gim...@gmail.com wrote: **The call-file I'm using is: Channel: SIP/08099...@outbound- calls MaxRetries: 3 WaitTime: 30 Set: LOCALSTATIONID=2 Set: LOCALHEADERINFO=T38 fax Set: T38CALL=1 Set: T38TXDETECT=yes CallerID: 2 Context: fax-out Extension: 2 priority:1 And upon further examination... don't put T38CALL in as a variable. It will cause the initial INVITE to only have T38. Leave it out and things should hopefully reinvite. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com www.asterisk.org http://www.asterisk.org ___ -- 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] dahdi wcb4xxp and fax
I have a couple of suggestions: Make sure that your timing configuration is correct in /etc/dahdi/system.conf (that it has a valid timing source). Also, you probably will probably want to use the half_full buffer policy, and set the number of buffers used to something reasonable, like 8, to ensure you don't have any transmit buffer underruns on the B410P. You shouldn't need more than that, since you're not trying to deal with clock slips or timing drift in this configuration. You may also try explicitly disabling the echo canceller. It seems that sometimes the CED tone detection (which disables the EC) takes a really long period of time to happen, and if it does, disabling the EC in the middle of the fax will usually cause a fax failure. Matthew Fredrickson Digium, Inc. Olivier wrote: 2009/2/25 stoffell stoff...@gmail.com mailto:stoff...@gmail.com Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) What about outgoing faxes ? Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1 machine. chan_dahdi.conf contains: faxdetect=both When receiving a fax call, hylafax (iaxmodem) answers the call after the obligatory wait of 3 seconds (fax detection) but to me it seems that echo cancellation is still being done. Theory is that any echo canceller hearing a 2100Hz fax signal would halt itself, so I wouldn't search in that direction first. Have you tried native 1.6 sendFax, receiveFax ? Maybe it would improve fax performance. Any pointers on this or workarounds? We're back to our old misdn setup for now ;) Here's some output from dahdi show channel 1 (the one that had the fax connection going), i cut out some non-related stuff : *CLI dahdi show channel 4 Signalling Type: ISDN BRI Point to Point Owner: DAHDI/4-1 Real: DAHDI/4-1 Callwait: None Threeway: None Confno: -1 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: yes Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Regards, stoffell ___ -- 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] Need some information on SS7 parameters
resea...@businesstz.com wrote: Can someone assist me on this please? Hello List I am setting up a small demo site using SS7 and one of the requirement is to be able to unhide the numbers and locate exact location of the caller (BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the parameters will be sent to the us. I just want to know how do read those information from the dialplan to be able to present them to the Agent It depends on what parameter this information is encoded inside. If you can find out the name of the parameter, we could probably answer your question. The likely answer is that we probably do not decode/expose this parameter to the dialplan at this time, but adding and exposing parameters is not a very hard thing to do. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which policy for ISDN BRI support in NT/PtMP ?
Olivier wrote: Hi, As you may know, these ISDN BRI features are very important here in Europe as ISDN Basic Rate Access is very popular among Small Medium Entreprises. I don't really know why but it seems that in many countries, default is to install small PBX using Point-to-Multipoint (PtMP) mode as opposed to Point-to-Point (PtP) which is the norm for PRI. So basically, in several countries, SME are equipped today with PBX connected with TE/PtMP interfaces to telco BRI lines. When we address those SME, my opinion is that it's very useful to be able to support any combination of TE/NT, PtP/PtMP modes. Latest 1.6 Asterisk and 1.4.8 Libpri introduced a new set of welcomed ISDN BRI features. Unfortunately, NT/PtMP is not available at this time, in latest Zaptel/Asterisk/Libpri. My question is what is the policy concerning NT/PtMP ? Is it really hard to extend Libpri to support this mode ? Or shall mISDN remain the way to go when NT/PtMP is needed ? Hey Olivier, I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. NT PTMP is very significantly different, in that you have to do much more from a TEI management perspective. Most people's needs that I saw were actually fulfilled in using either NT or TE PTP or TE PTMP, since they were interfacing with PBXs or using TE-PTMP trunks from the telephone network to provide voice trunks for Asterisk. Right now, I would not preclude the possibility that NT-PTMP support might be added, but I could not give you a concrete time at which it will be done, since it will probably require some significant internal changes in libpri. To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which policy for ISDN BRI support in NT/PtMP ?
Patrick wrote: Matthew Fredrickson wrote: [snip] I actually was the one that did a lot the work in adding the BRI support to libpri/chan_dahdi. [snip] To answer your final question, for now, if you need NT-PTMP mode, you should use mISDN. Hi Matthew, Is there a BRI status document? I'm asking because it's not clear to me There release logs that are made whenever we make a new release of libpri or Asterisk which contain information about development in this area. if I need mISDN or that Digium (you) has developed native support for the B410P card BRI card in zaptel/dahdi/libpri. If there's native support for BRI, which version(s) of zaptel/dahdi/libpri would I need to install to test this? You must have the most current version of DAHDI, libpri-1.4, and a version of Asterisk-1.6. Matthew Fredrickson Digium, inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as MGCP client
Bob Pierce wrote: Has there been any work done on using Asterisk as an MGCP client? Nope :-( Still a no go. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Full Duplex
Matt Riddell wrote: On 18/11/2008 9:46 a.m., Matthew Fredrickson wrote: Singer X.J. Wang wrote: We've had the same issue. For calls that go between a SIP connection (desktop phones) and Zaptel connections, there was a lot of problems with half duplex. We switched from the Digium card to the Sangoma card and the problem went away. Just for the record, he said that it happened regardless of protocol (IP to IP calls do not use the card based echo cancellers). Sorry for the problem you had. However, I think that if you use a current version of our echo canceller board, you will find your issues resolved. In fact, for a significant number of Digium's boards, Sangoma uses the exact same hardware echo canceller. A few months ago, I had a similar problem and needed to pass: vpmnlptype=4 vpmnlpmaxsupp=11 to resolve it. If I upgraded zaptel would this be fixed? In a newer version of the firmware, it's very likely, although you would have to talk to technical support directly about it right now to get the updated version. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card Noice issue
Bipin wrote: Hello all, I am facing as serious problem when running asterisk in HP server.We are developing application to make the outbound calls in PRI lines .We normally uses IBM machine as our servers ,and it was working fine for all installation.For the cost reduction we this time tried with HP server. Model(HP proliant ml110). When we make the calls the there is a lots of disturbance in the sound even if we make a single calls the issue persist .I found in google that these issue normally comes by the load or by the line or by the IRQ . As in my case i am making a single call the 1 st case wont occur here.Also i tested it with one smoothly working E1 to the same card and still the problem came.so I guess the problem is with IRQ. But when i tried it with a normal PC with pendium 4 processor it was working fine. My question is whether the Digium card had any hardware compatibility issue with HP proliant ml110 server.Why the sound has issue in HP server when it working fine in a normal pC with pendium processor...?? When i switched to Asterisk now it is very much ok. can any body explain why it have when using with ubuntu??? I would definitely report your issue to Digium technical support so that we can correct whatever issue this might be. It might be something very simple causing the problem (like X running or something like that) which is making the system's interrupt latency increase to something at an unreasonable level for Zaptel to operate properly. In any case, it's possible that it's a very quick problem to fix and that technical support will be able to help you with. Also, make sure before you try calling to verify that it's not an OpenVox card. I have had customers who thought they were sold a Digium card, but were in fact sold an OpenVox card which did not perform as well as would be expected from our cards, causing some grief and confusion for them. It can be confusing because they use the Digium driver and look like a Digium card to the driver. --- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
John Todd wrote: On Nov 11, 2008, at 3:44 PM, Steve Murphy wrote: On Tue, 2008-11-11 at 16:11 -0700, Wilton Helm wrote: I'm a bit puzzled, also, having implemented ulaw and alaw in an embedded application. Each can be done with a 16 Kbyte table in about 0 time with no errors. There are probably tricks that will cut the table down by 2 or 4 X for a small cost in CPU cycles. The inverse requires 256 16 bit words. I thought ulaw and alaw were pretty much no brainers. I don't know of any gottchas. Why anyone with more that a few K bytes of total system memory would even consider anything other than a lookup table is beyond me. Actually, with the way caching is done on nearly all modern processors, it is debatable whether or not a look up table is the optimal way to do the conversion, at least on such a simple codec such as ulaw or alaw. In fact, the amount of time it takes to fetch memory from a cache miss can easily ruin the single element lookup performance in a look up table. And if you have large tables (such as in the linear to ulaw or alaw table), the tradeoff of having to service a cache miss versus a few cached instructions executing a native CPU clock speed makes it almost a no brainer (IMHO). You'll pay a cache miss on the first time your run the routine, but the instructions running the routine will take up much less CPU cache space than the look up tables, increasing the likelihood of them being evicted (whereas the lookup table, taking up a lot more space, has a much better chance of causing a cache miss whenever you access). Obviously, if you're running on a CPU with no cache, a look up table is a good way to do it. I'm just saying that very few processors that are running Asterisk are running it on processors without processor caches. Matthew Fredrickson Digium, Inc. Wilton Wilton-- AFAIK, the current algorithms (old new) are indeed table lookup. It wouldn't hurt for you to do a code review on them, you might be able to improve them...! murf For those of you interested in a slightly longer discussion here, there is discussion (Nov 14) on the voip-users-conference about this and many other things: http://www.talkshoe.com/talkshoe/web/talkCast.jsp?masterId=22622cmd=tc JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- 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] Full Duplex
Singer X.J. Wang wrote: We've had the same issue. For calls that go between a SIP connection (desktop phones) and Zaptel connections, there was a lot of problems with half duplex. We switched from the Digium card to the Sangoma card and the problem went away. Just for the record, he said that it happened regardless of protocol (IP to IP calls do not use the card based echo cancellers). Sorry for the problem you had. However, I think that if you use a current version of our echo canceller board, you will find your issues resolved. In fact, for a significant number of Digium's boards, Sangoma uses the exact same hardware echo canceller. Matthew Fredrickson Digium, Inc. Doug Lytle wrote: Ken Williams wrote: We’ve had an issue since we went live nearly two years ago on Asterisk where people complain about not being able to talk while someone else is talking. I had assumed for a very long time this was because of the phones we went live with (Grandstream GXP-2000’s) and for the longest time I Sound like your echo canceller is set to aggressive. Make it like a walky-talky. From the archives, January 2006: On Friday 20 January 2006 15:36, Ronald Hartmann wrote: Anyone know if it is possible to control how aggressively the Aggressive mode behaves. No. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie talkie behaviour of the Aggressive mode. The agressive canceller is agressive because it is designed to turn your voice channel into a half-duplex (walkie-talkie) communications channel. You can't have a half half duplex situation. :-) Have you tried recent SVN trunk with the MG2 echo canceller? I have found that to be the absolute best to date. -A. Doug -- *Singer Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free:(877) 798-4426 x298 Fax: (613) 565-8710 Email:[EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Yahoo:pythianwang AIM: pythianwang ICQ: 201253 Gadu-Gadu:6817795 Tencent QQ: 858310404 ___ -- 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] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Benny Amorsen wrote: Matthew Fredrickson [EMAIL PROTECTED] writes: Actually, with the way caching is done on nearly all modern processors, it is debatable whether or not a look up table is the optimal way to do the conversion, at least on such a simple codec such as ulaw or alaw. In fact, the amount of time it takes to fetch memory from a cache miss can easily ruin the single element lookup performance in a look up table. If the compiler is clever enough, you can embed a small lookup table in the instruction stream. Instruction prefecting will automatically ensure the page is in I-cache, and even on most processors which can't read from I-cache the table will be in 2nd-level cache. Low-level optimizations like these are often dependent on processor architecture though. This is very true. Mostly wanted to make sure that people knew that cache miss penalties can be more of a slow down (and in fact will be for a simple thing like a lin-to-mu) on a big, multi page table like in a lin-to-mu lookup than simply executing the instructions from I-cache (which are much more likely to not cause a miss due to the small number of instructions involved). Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Steve Underwood wrote: Matthew Fredrickson wrote: Actually, with the way caching is done on nearly all modern processors, it is debatable whether or not a look up table is the optimal way to do the conversion, at least on such a simple codec such as ulaw or alaw. In fact, the amount of time it takes to fetch memory from a cache miss can easily ruin the single element lookup performance in a look up table. And if you have large tables (such as in the linear to ulaw or alaw table), the tradeoff of having to service a cache miss versus a few cached instructions executing a native CPU clock speed makes it almost a no brainer (IMHO). You'll pay a cache miss on the first time your run the routine, but the instructions running the routine will take up much less CPU cache space than the look up tables, increasing the likelihood of them being evicted (whereas the lookup table, taking up a lot more space, has a much better chance of causing a cache miss whenever you access). Obviously, if you're running on a CPU with no cache, a look up table is a good way to do it. I'm just saying that very few processors that are running Asterisk are running it on processors without processor caches. Matthew Fredrickson Digium, Inc. In spandsp I do the G.711 conversions algorithmically. Most modern processors have a where is the top 1 instruction, and that reduces the calculations to something very fast. When I first did this it was a lot slower than a lookup if I tested it on its own, but faster in a real workload where the cache was working hard. That was in the days of 256k caches, though. Now the latest Intels have 12M the picture may be different. That 12M is L3 cache, which is a lot slower than the small L1 cache, but I suspect it make mean the lookup approach is as good as calculation with any workload. That's a pretty good point too. A lot of this is speculation until an actual workload is put through the mix. I would suspect though that you're more likely to be faster on a larger range of processors in use at the moment (the bulk my guess wouldn't have 12 MB L3 caches) with the algorithmic approach, like you mentioned. And if it's just a few instructions, it quite possibly could be faster than a combined L1 and L2 cache miss (IMHO :-) ). Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use the NEW ulaw/alaw codecs (slower, but cleaner)
Steve Underwood wrote: Matthew Fredrickson wrote: Actually, with the way caching is done on nearly all modern processors, it is debatable whether or not a look up table is the optimal way to do the conversion, at least on such a simple codec such as ulaw or alaw. In fact, the amount of time it takes to fetch memory from a cache miss can easily ruin the single element lookup performance in a look up table. And if you have large tables (such as in the linear to ulaw or alaw table), the tradeoff of having to service a cache miss versus a few cached instructions executing a native CPU clock speed makes it almost a no brainer (IMHO). You'll pay a cache miss on the first time your run the routine, but the instructions running the routine will take up much less CPU cache space than the look up tables, increasing the likelihood of them being evicted (whereas the lookup table, taking up a lot more space, has a much better chance of causing a cache miss whenever you access). Obviously, if you're running on a CPU with no cache, a look up table is a good way to do it. I'm just saying that very few processors that are running Asterisk are running it on processors without processor caches. Matthew Fredrickson Digium, Inc. In spandsp I do the G.711 conversions algorithmically. Most modern processors have a where is the top 1 instruction, and that reduces the calculations to something very fast. When I first did this it was a lot slower than a lookup if I tested it on its own, but faster in a real workload where the cache was working hard. That was in the days of 256k caches, though. Now the latest Intels have 12M the picture may be different. That 12M is L3 cache, which is a lot slower than the small L1 cache, but I suspect it make mean the lookup approach is as good as calculation with any workload. Or (in continuation of my email I just sent), the better chances of it fitting in L1 (or event L2) cache, the quicker it's going to run :-) Maybe that's a better way to look at it. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6 Production ready??
Sebastian Gutierrez wrote: What Hardware? For that performance? It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM. Oh yeah, those numbers indicate averaging over 110,000 calls per day (the ones I posted below) :-) -- Matthew Fredrickosn Digium, Inc. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] 1.6 Production ready??
Sebastian Gutierrez wrote: What Hardware? For that performance? It's a dual core 1.8 GHz Opteron with 2 TE420P cards and 4 GB of RAM. Oh yeah, those numbers indicate averaging over 110,000 calls per day (the ones I posted below) :-) -- Matthew Fredrickosn Digium, Inc. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew Fredrickson Enviado el: Friday, November 07, 2008 3:18 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] 1.6 Production ready?? Steve Totaro wrote: On Fri, Nov 7, 2008 at 11:50 AM, Matthew Fredrickson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Sebastian Gutierrez wrote: Anyone is using 1.6 in production?? Is it ready? I have a number of people using 1.6 in production doing SS7-SIP, SS7-IAX, and SS7-ISDN gatewaying. One example (doing SS7-IAX): System uptime: 10 weeks, 6 days, 21 hours, 35 minutes, 45 seconds 8617029 calls processed --- Matthew Fredrickson Digium, Inc. EEEK IAX!! Do you use IAX for a reason? Is it because Asterisk does not setup SIP calls very well? Just curious. The customer chose to use IAX. It has been working very well for him. Impressive, but very purpose specific. Do you only load a couple of modules? Full suite of modules, although it is not using most of them. I did specifically mention in the original message that it was primarily being used as a gateway machine. I think the question was more along the lines of what Asterisk was meant to be, a feature rich PBX. Maybe.. or maybe not. In any case, this is some specific data that someone can use about 1.6's performance. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] ISDN PRI Caller ID problem
A.R. Nasir Qureshi wrote: Dear All, I am trying to setup an ISDN line from local telco on a digium card. The problem I am facing is that I am not getting any caller id from the telco. They say that they have enabled caller id. Tell them they are wrong. There is no calling party number IE in that SETUP message below. :-) Matthew Fredrickson Digium, Inc. Please help me out. My zapata.conf [trunkgroups] [channels] context=pstnincoming pridialplan=local prilocaldialplan=local usecallerid=yes cidsignalling=v23 cidstart=ring hidecallerid=no callwaiting=no usecallingpres=yes sendcalleridafter=1 echocancel=no echocancelwhenbridged=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived busydetect=no busycount=6 callprogress=no faxdetect=incoming switchtype = national signalling = pri_cpe group = 1 channel = 1-15,17-31 channel = 32-46,48-62 The information I get from using pri intense debug span 1 is: [ 02 01 16 9c 08 02 15 01 05 04 03 80 90 a3 18 03 a1 83 81 70 08 c1 34 33 39 32 38 34 32 a1 ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 011 0: 0 N(R): 078 P: 0 26 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 77 to (but not including) 78 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=26 Call Ref: len= 2 (reference 5377/0x1501) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [70 08 c1 34 33 39 32 38 34 32] Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4392842' ] [a1] Sending Complete (len= 1) -- Making new call for cr 5377 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 161 (cs0, Sending Complete) q931.c:3509 q931_receive: call 5377 on channel 1 enters state 6 (Call Present) Sending Receiver Ready (12) ___ -- 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 aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
Steve Totaro wrote: On Thu, Oct 9, 2008 at 10:32 PM, sean darcy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Remco Barendse wrote: The information (or lack of it) on upgrading from zaptel to that @*^QW%^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make uninstall and make remove in the zaptel directory which just generates errors and does nothing (on zaptel 12.1). Then i install dahdi-linux and dahdi-tools and i want to start configuring it, so i am trying dahdi_genconf like the docs suggested which generates this really helpful error message : /usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No such file or directory Also the config files and everything are much more complicated for dahdi than they were for zaptel There was some nice documentation and examples on how to get started with configuring certain devices with zaptel on the digium page, for my TDM11B they only mention zaptel. Did anyone even try this? It'll work. But it's not easy. I didn't find dahdi_genconf helpful. Post your /etc/dahdi/system.conf ( the analogue of zaptel.conf ) and /etc/asterisk/chan_dahdi.conf ( analogue of zapata.conf ). With some help, you'll fix this. sean Total hindsight and thinking as a user, but the initial explanation of DAHDI came out because someone put something out there premature and someone noticed that Zaptel was being replaced by DAHDI. The party line explanation from Digium was that someone owned the rights to the zaptel name. A calling card dealer who had been very nice to allow Digium to continue using the Zaptel name but was at his end, so hence the name change. This *is* the correct reason. Not sure I totally buy that but whatever, my thought was it was to remove any rights or credits from the Zapata Telephony Project and Jim Dixon. Digium could control DAHDI exactly the way it controls Asterisk, Jim's name is still on the source code, and still intentionally is there. Please don't jump to any rash conclusions. You can certainly still use Zaptel as Zaptel if you'd like. We were forced to change it due to the name related issues that have been mentioned. We're just grateful that the other party that brought the issue up has been so patient since it has taken so long. It has been a bit of a rocky road with some of the new features that were put it into it, but, any time you rewrite code or do something new, there's always going to be a period of shaking out of unforeseen bugs. Sorry if you have had any trouble. The name change and related efforts have been just as hard on us as developers as it has been on people that use it. -- Matthew Fredrickson Software/Hardware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI E1 Inbound calls hangup with busy after a few seconds
Daniel Johnson wrote: Hi, I have a 10 line PRI E1 ISDN service from AAPT. Connected to Asterisk 1.4 via a Digium TE121P. All oubound calls work fine. Inbound works only if I Dial a SIP phone directly or as the first step. This phone MUST NOT be busy or else the call will fail. It would appear that your zapata.conf is incorrect. One of your problems (and maybe all of them) is that it looks like you used the spanmap and trunkgroup way to setup your PRI. This is *ONLY* supposed to be used for NFAS PRIs (where you have multiple trunks per PRI). There is some tricky code in there which, when configuring using those parameters, causes it to send the channel ID in an NFAS friendly manner (which most non NFAS configured switches do not appreciate). Instead, comment out any trunkgroups or spanmaps you may have setup in zapata.conf, and do as follows: signalling=pri_ ; where pri_ is either pri_net or pri_cpe channel=1-23 ; or whatever your channels are associated with the PRI. Matthew Fredrickson Digium, Inc. eg. This works: [from-pstn] exten = _34397333,1,Dial(SIP/511,30) exten = _34397333,n,Hangup This also works: exten = _34397333,1,Dial(SIP/511,2) exten = _34397333,n,GoTo(ConfMe,s,1) exten = _34397333,n,Hangup This does not work: exten = _34397333,1,GoTo(ConfMe,s,1) exten = _34397333,n,Hangup [ConfMe] exten = s,1,Answer exten = s,n,MeetMe(9000|crM|) exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Below I have included the PRI debug output for a successful and failed inbound call. But first I have done a break down of the messages received for both failed and successful calls. Failed: Message type: SETUP (5) Message type: CALL PROCEEDING (2) Message type: CONNECT (7) Message type: STATUS (125) Message type: STATUS (125) Message type: SETUP (5) Message type: DISCONNECT (69) Message type: RELEASE (77) Message type: RELEASE COMPLETE (90) Successful: Message type: SETUP (5) Message type: CALL PROCEEDING (2) Message type: ALERTING (1) Message type: STATUS (125) Message type: CONNECT (7) Message type: CONNECT ACKNOWLEDGE (15) Message type: STATUS (125) And now the call is fully connected... Now I have no experience with PRI ISDN. However it seems to be that the ALERTING message is important and the Dial(SIP/511,30) is forcing this response as it starts to ring. Why Answer() does not do this I think is my problem... however this is probably because of a config issue. Or am I missing something? Could it be that ISDN service has not been setup correctly? I have called AAPT and their tech ran test and think it is all configured correctly on their side. Thank you for your help. zaptel.conf loadzone=au defaultzone=au # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) B8ZS/ESF RED span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17,31 dchan=16 -- zapata.conf [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] ; Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) B8ZS/ESF RED group=1 context=from-pstn switchtype=euroisdn signalling=pri_cpe channel = 1-10 group=1 context=default PRI debug output for a failed inbound: Protocol Discriminator: Q.931 (8) len=45 Call Ref: len= 2 (reference 3687/0xE67) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) User information layer 1: A-Law (35) [18 03 a1 83 87] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 7 ] [6c 0c 21 83 30 37 33 33 38 37 35 35 35 35] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '073387' ] [70 09 c1 33 34 33 39 37 33 30 30] Called Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '34397300' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) [a1]sk*CLI Sending Complete (len= 1) -- Making new call for cr 3687 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) -- Processing IE 161 (cs0, Sending Complete) q931.c:3509 q931_receive: call 3687 on channel 7 enters state 6 (Call Present) q931.c:2774 q931_call_proceeding: call 3687 on channel 7 enters state 9 (Incoming Call
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Jay R. Ashworth wrote:
On Mon, Sep 08, 2008 at 11:28:13AM -0500, Matthew Fredrickson wrote:
For DMS100's version of TBCT, called RLT, one leg *must* be inbound and
the other *must* be outbound. No other combination is going to work.
This is explicitly mentioned in the protocol in RLT.
Ok.
Just found this in my archive.
Matt: should I assume that this implies that if my switch is provisioned
for NI2, and my Asterisk is set to DMS, that things aren't going to work
well at all? :-) (Outbound calls, FWIW, seem to work fine like that...)
Probably not. You can obviously try this out, but don't be surprised if
this doesn't work. You usually want to have your switchtype (which
likewise sets the version of TBCT which is used) set to the same thing
that the other end is provisioned to be.
Ok. I've run a simple test:
exten = 727xxx,1,Dial(${TRUNKY}/727yyy,,r)
exten = 727xxx,2,Hangup
Where TRUNKY is a group that points to the same T-1 on which the calls
are coming in.
And what I get is:
-- Accepting call from '727zzz' to '727xxx' on channel 0/1, span 4
-- Executing Dial(Zap/73-1, Zap/g3/727yyy||r) in new stack
-- Requested transfer capability: 0x10 - 3K1AUDIO
-- Called g3/7276471274
-- Zap/74-1 is proceeding passing it to Zap/73-1
-- Zap/74-1 is ringing
-- Zap/74-1 answered Zap/73-1
-- Attempting native bridge of Zap/73-1 and Zap/74-1
-- Channel 0/1, span 4 got hangup request, cause 16
-- Hungup 'Zap/74-1'
== Spawn extension (default, 727xxx, 1) exited non-zero on 'Zap/73-1'
(I think I got all those numbers sanitized properly.)
And yes, the call went through, and had the CNID of the originating
phone, as I want.
So, since I can't tell from the logs -- no timestamps -- I have to guess
from when the messages show up, but I can't tell if the attempted native
bridge is *succeeding*. How would I know that it had? We do
*successful* ones in other contexts, and I don't recall seeing a
'success' message on those.
Will I actually need to do PRI debug on that span to tell?
Or will seeing hangup messages while I'm still talking be the solution?
Seeing hangup messages on the console while the audio path remains
indicates success :-)
--
Matthew Fredrickson
Digium, Inc.
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Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Jay R. Ashworth wrote: On Fri, Sep 12, 2008 at 10:56:40AM -0500, Matthew Fredrickson wrote: Will I actually need to do PRI debug on that span to tell? Or will seeing hangup messages while I'm still talking be the solution? Seeing hangup messages on the console while the audio path remains indicates success :-) Then, as I suspected, I'm failing. I need to confirm that it's actually provisioned with the carrier, and which switchtype I'm really on. Can *you* confirm, off hand, that 1.2 would do TBCT at *all*? Someone on IRC thinks it wouldn't. It will only attempt it for DMS100 switchtype. You must have 1.4 libpri for any other switchtype. Matthew Fredrickson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Jay R. Ashworth wrote: On Fri, Aug 15, 2008 at 03:03:23PM -0500, Matthew Fredrickson wrote: Let me clarify some of this. Under no circumstances can Asterisk receive a TBCT request. We just ignore them. We can initiate them however. There are different TBCT implementations, dependent on which switch type is used, with different restrictions associated with each switch type selected. For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any combination of inbound and/or outbound channels (one inbound/one outbound, two inbound, two outbound) and transfer them to the upstream switch. The protocol doesn't care. For DMS100's version of TBCT, called RLT, one leg *must* be inbound and the other *must* be outbound. No other combination is going to work. This is explicitly mentioned in the protocol in RLT. Ok. Just found this in my archive. Matt: should I assume that this implies that if my switch is provisioned for NI2, and my Asterisk is set to DMS, that things aren't going to work well at all? :-) (Outbound calls, FWIW, seem to work fine like that...) Probably not. You can obviously try this out, but don't be surprised if this doesn't work. You usually want to have your switchtype (which likewise sets the version of TBCT which is used) set to the same thing that the other end is provisioned to be. Matthew Fredrickson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DSS1 vs SS7
Kevin P. Fleming wrote: Alex Balashov wrote: Some carriers now do offer private SS7 instead of ISDN. But there is absolutely no reason why you should be doing this with Asterisk. Asterisk-SS7 is quite tenuous at best. Unless you have some specific reason to be using it, don't. Actually, SS7 support in Asterisk 1.6.0 appears to be quite solid, and it is being used in a quite a number of production deployments. Thanks for the plug Kevin! :-) Yeah, actually, if you guys want to know more there's an asterisk-ss7 mailing list. Asterisk-1.6.0 with libss7 is being used in many successful and high traffic installations around the world. The current record (that I have been told of) is an installation doing over 100,000 calls per day. So try to beat that ;-) Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Tilghman Lesher wrote: On Friday 15 August 2008 13:45:11 Jay R. Ashworth wrote: On Fri, Aug 15, 2008 at 02:37:46PM -0400, Matt Florell wrote: Most carrier sales people don't know what TBCT is unfortunately, and even if a carrier is capable of doing it, it is a possiblity that not all of their equipment is capable of doing it. One client of mine tried to get TBCT working across all 16 of their PRIs(all on the same carrier) and it only worked on 4 of them, supposedly because not all of the telco equipment was capable of the feature. I expect to fight this battle, yes. :-) This actually depends on the kind of PRI service you have. For instance with DMS100 circuits you can only do TBCT with calls that come in to your circuit, not with outgoing calls. As for connecting two incoming calls, since that is not possible in Asterisk(to natively bridge two incoming calls together) I can't see how you would get that to work even if it is possible in TBCT. To be more clear, what I'm after is to have *someone else besides me* place calls out their PRI, and then TBCT those placed calls to my DN. By the time the calls get to me, they should just be standard phone calls. So I expect the call-placing-party to need TBCT, but not me. I believe that only DMS100, NI2 and 5ESS PRI signalling protocals are capable of TBCT with the current zaptel code-base. Also, the two B channels involved in the TBCT have to use the same D channel. And I'm probably not concerned with whether Asterisk can deal with TBCT, because Asterisk probably won't be involved at that stage; just once the call's transferred to me. But before I inquire of said second party whether they *can* do that, I wanted to confirm it was possible. 2BCT works when the telco originates the call and Asterisk is hairpinning the call back out the same PRI circuit. However, Asterisk does not support the opposite direction. That is, a call originated from Asterisk that comes back in via the same PRI circuit cannot be 2BCT. I'm not certain whether this is a limitation of Asterisk alone or of the protocol, but it cannot be done. Similarly, Asterisk cannot complete a 2BCT request, if Asterisk is on the NET side of the PRI circuit. That might could be added in the future, but it is not supported now. So in summary, Asterisk can request 2BCT, but it cannot perform a 2BCT if requested from the other side. Let me clarify some of this. Under no circumstances can Asterisk receive a TBCT request. We just ignore them. We can initiate them however. There are different TBCT implementations, dependent on which switch type is used, with different restrictions associated with each switch type selected. For true TBCT (on switchtypes of NI2 and 5ESS, AFAIK), you can have any combination of inbound and/or outbound channels (one inbound/one outbound, two inbound, two outbound) and transfer them to the upstream switch. The protocol doesn't care. For DMS100's version of TBCT, called RLT, one leg *must* be inbound and the other *must* be outbound. No other combination is going to work. This is explicitly mentioned in the protocol in RLT. Hope that helps a bit. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
emist wrote: My best guess from looking at that is that its a driver bug. The last thing that happens before the lockup seems to be an ioctl call to the device. That was a bug that should have been resolved by 1.4.11 (he subsequently updated and it was resolved). Matthew Fredrickson Digium, Inc Hope it helps, Igor H. Lee, John (Sydney) wrote: This time, I am trying to remotely install Asterisk in China. I was told that an E1 line has been installed and so I plug it into port 1 of a TE412P. On the box, first of all, I just installed Zaptel 1.4.10.1. # service zaptel restart Unloading zaptel hardware drivers:ERROR: Module zaptel is in use . Loading zaptel framework: [ OK ] Waiting for zap to come online...OK Loading zaptel hardware modules: tor2. wct4xxp. wcte12xp. wct1xxp. wcte11xp. wctdm24xxp. wcfxo. wctdm. wcusb. Running ztcfg: [ OK ] # vi zaptel.conf [...] span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 *** However, I received a red alarm in zttool and the LED on the TE412P card is also red. *** I have made sure that the jumper is closed for port 1 on the TE412P card and so it could not be the jumper problem. ### Because this is the first time I install Asterisk in China and I was wondering if their E1 is different from the Euro E1. ### However, I went into dmesg and I discovered the following. ### Could it really be a zaptel bug? I saw on a similar few on the digium bug list but I cannot be 100% sure. Any thoughts? About to enter spanconfig! Done with spanconfig! Registered tone zone 33 (China) About to enter startup! TE4XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct4xxp: Setting yellow alarm on span 1 timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 128 channels BUG: soft lockup - CPU#2 stuck for 16s! [ztcfg:4681] Pid: 4681, comm:ztcfg EIP: 0060:[f8cba1df] CPU: 2 EIP is at init_vpm450m+0x32d/0x34a [wct4xxp] EFLAGS: 0286Tainted: G (2.6.18-92.1.6.el5 #1) EAX: EBX: f76ae8f0 ECX: 0019 EDX: ESI: f7997400 EDI: 0286 EBP: f76838c0 DS: 007b ES: 007b CR0: 8005003b CR2: b7f01007 CR3: 37bc1000 CR4: 06d0 [f8ca1b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8ca5ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042609c] release_console_sem+0x17e/0x1b8 [c046d53a] cache_alloc_refill+0x14b/0x450 [f8956f61] zt_ioctl+0x273/0x144f [zaptel] [c04d7d45] generic_make_request+0x248/0x258 [c045ae3c] __do_page_cache_readahead+0x69/0x1c6 [c0484a5b] __d_lookup+0x98/0xdb [c047c110] do_lookup+0x53/0x166 [c047e7e4] do_path_lookup+0x20e/0x25e [c047c389] permission+0xa2/0xb5 [c04e2d06] kobject_get+0xf/0x13 [c046f7fa] __dentry_open+0xea/0x1ab [c046f91f] nameidata_to_filp+0x19/0x28 [c046f959] do_filp_open+0x2b/0x31 [c048029b] do_ioctl+0x47/0x5d [c04804fb] vfs_ioctl+0x24a/0x25c [c0471bbe] __fput+0x13f/0x167 [c0480555] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Completed startup! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie in China: Red alaram in Zaptel for E1
Lee, John (Sydney) wrote: The test for that is simple: head -n 1 /proc/zaptel/* Let's look at all four spans. Not just the first one. Thanks Tzafrir. # head -n 1 /proc/zaptel/* == /proc/zaptel/1 == Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/ RED == /proc/zaptel/2 == Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 == /proc/zaptel/3 == Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 == /proc/zaptel/4 == Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 So I am quite sure that port 1 is plugged in properly. As I am dealing with telecom in China, I think I might have stepped onto the MFC R/2 bombshell but I have no idea whether the signalling is ISDN or R2. I tried the suggestion on http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 but the red alarm is still on. If it is really R2, then maybe I need to buy an E100P card instead of TE412P. No, you should be fine with a TE412. Just make sure that your line is plugged in correctly and your span= line is correct for the line settings. Matthew Fredrickson Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disconnect on PRI ignored?
Alexander Zielke wrote: Hi List, i recently set up a system with a TE410P. Everything works, except that disconnects don't seem to be processed. Here is what i get: -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 23819/0x5D0B) (Originator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3779 q931_receive: call 23819 on channel 6 enters state 12 (Disconnect Indication) -- SIP/2025-08245ac8 is ringing -- SIP/2025-08245ac8 is ringing ... I just made a call from the outside to a local SIP-Phone, but when the outside call hangs up, the Phone keeps ringing. The call will only hangup, if i take the call, or wait for the call to time out. The only similar thing i found is the bug at http://bugs.digium.com/view.php?id=9588, but that seems fixed in 1.4.21.1. Did anyone else experienced something like that? If you are using libpri-1.4.4, you should either downgrade to 1.4.3 or upgrade to 1.4.5. A new default behavior was introduced in 1.4.4 (which should have been optional, not default) which causes a channel to be left open until the RELEASE timer expires when a DISCONNECT is received with Inband progress information avaiable. Matthew Fredrickson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran fxotune to tune the lines. fxotune.conf ended up looking exactly the same as before the change. Since I was expecting/hopping to see a change but did not I switched everything back to the way it was. Is there a way to test the lines, using a multi-meter maybe, to tell if the tip and ring are correct or reversed? After putting things back I reran fxotune to get the verbose output. It, foxtune.out.gz, is attached. fxotune seems to have had a better time with line 7 during this run. fxotune.conf now contains: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=4,0,0,0,0,0,0,0,0 8=7,255,251,251,2,255,255,1,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 I tried calling directly into the lines above and it seems lines 5,6,8 have much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to the following and reloaded (fxotune -s) it: 5=5,0,0,0,0,0,0,0,0 6=5,0,0,0,0,0,0,0,0 7=4,0,0,0,0,0,0,0,0 8=5,0,0,0,0,0,0,0,0 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 Unless I am just spacing out the echo on 5,6,8 seems less now. I really have no idea what is going on. Ok, I looked at the output of you running fxotune. Basically, the lines that have numbers in them besides 0 (after the first two terms x=y,...) are the complex line simulation line models. The output you gave me demonstrated that they gave the best return loss characteristics using the built in test frequencies. It's possible that your setup is not performing well with these line models, which is why you might notice less echo using the second set of settings you listed above. Which echo canceller are you using with this, by the way? (Hardware, software, if software, which software echo canceller). Matthew Fredrickson John On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- 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 ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list
Re: [asterisk-users] fxotune vs rxgain/txgain
Noah Miller wrote: Hi Matthew - These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. Thanks for your responses! I actually didn't realize there are hardware gain settings available for wctdm24xxp (is there any documentation on this? I can't seem to find any). I assume the hardware gains default to 0 if left unset? Correct. They are set as module parameters, and actually only apply to fxo modules. Just two more questions: 1) I think we were experiencing ECFO with an rxgain setting of +10db (after having balanced the hybrid using fxotune). I'm guessing this is because that rxgain value amplifies the echo a bit too much. I know this is a bit of a loaded question, but is there a certain range of values for rxgain/txgain that we should stay within in order to avoid exacerbating any echo issues? I couldn't give you exact numbers off the top of my head. It's not hard to notice though if it's happening :-) 2) Are rxgain/txgain values applied before or after hardware echo cancellation? rxgain is pre-hardware echo canceller and txgain is post hardware echo canceller. (zapata.conf rxgain and txgain). -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: I switch the wires in lines 5-8 (i.e. reversed tip and ring) and reran fxotune to tune the lines. fxotune.conf ended up looking exactly the same as before the change. Since I was expecting/hopping to see a change but did not I switched everything back to the way it was. Is there a way to test the lines, using a multi-meter maybe, to tell if the tip and ring are correct or reversed? After putting things back I reran fxotune to get the verbose output. It, foxtune.out.gz, is attached. fxotune seems to have had a better time with It seems that one way or another the attachment didn't go through. Can you email the tarball to me directly or post it to a website? Thanks, Matthew Fredrickson line 7 during this run. fxotune.conf now contains: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=4,0,0,0,0,0,0,0,0 8=7,255,251,251,2,255,255,1,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 I tried calling directly into the lines above and it seems lines 5,6,8 have much more echo than lines 7,9,10. So just for fun I edited fxotune.conf to the following and reloaded (fxotune -s) it: 5=5,0,0,0,0,0,0,0,0 6=5,0,0,0,0,0,0,0,0 7=4,0,0,0,0,0,0,0,0 8=5,0,0,0,0,0,0,0,0 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 Unless I am just spacing out the echo on 5,6,8 seems less now. I really have no idea what is going on. John On Fri, Jun 6, 2008 at 1:31 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- 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 ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune vs rxgain/txgain
Noah Miller wrote: Well, that clears it up a little. I think where I get confused is that sometimes using fxotune is called balancing the hybrid and some times using ztmonitor and adjusting the txgain/rgain settings is called balancing the hybrid. Perhaps they both try to achieve the same goal, but through different means? Not quite. Gain adjustment affects volume levels of the respective direction you are adjusting (echo and all). Balancing the hybrid via fxotune attempts to balance the hybrid in a manner so that the hybrid will remove as much of the echo as possible. This leads me to my other question - Are these two techniques mutually exclusive? In some posts from Matthew Frederickson, it seems that they are, and that if you use fxotune, you should set your gains back to zero. Some other people seem to suggest using both fxotune and adjusting gain levels. I note that Stephen Bosch asked just this question some time back, and nobody was able to answer him. These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fxotune question
John Morey wrote: Tilghman, Thanks for the pointer. I'll check this tomorrow and let you know. Also, I would like to see the output without the -d flag and with the -v flag. This will output a lot of data (the echo ratio for every possible coefficient setting it has tried per port). Matthew Fredrickson John On Wed, Jun 4, 2008 at 11:18 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 04 June 2008 22:02:19 John Morey wrote: Hello, I've run fxotune at different times but continue to get what seem to be strange numbers in /etc/fxotune.conf. It ends up with: 5=7,255,251,251,2,255,255,1,255 6=7,255,251,251,2,255,255,1,255 7=7,255,251,251,2,255,255,1,255 8=9,2,250,253,4,252,0,255,255 9=4,0,0,0,0,0,0,0,0 10=5,0,0,0,0,0,0,0,0 11=0,0,0,0,0,0,0,0,0 12=0,0,0,0,0,0,0,0,0 ports 5-10 have lines hooked up to them. The first four lines seem strange when compaired to what others have posted and what ports 9 and 10 have. Also if I'm reading things right my echo ratios seem to be very high. Running fxotune -d -b 5 -w 1004 gives the following: Dumping module /dev/zap/5 echo ratio = 0.1759 (1960.0 / 11145.0) Which I read to be over 17%. This seems crazy. Am I reading this right? Where should I start to look for problems? You might check to see if the tip and ring are reversed in your wiring. That can frequently cause weird echo problems. -- 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 ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 over ISDN PRI/BRI
Simon Hyde wrote: Hi, G.722 is heavily used by Broadcasters worldwide for wideband voice communications over ISDN. I'd like to be able to receive these G.722 over ISDN calls into an Asterisk exchange (with mostly a view to routing the calls to a Voicemail box where material can be recorded). I have been examining source code for the 3 different ISDN Channels in Asterisk and they all seem to be hard- codec to aLaw/uLaw G.711. It looks as though chan_capi *might* support bridging of G.722 data from one ISDN port to another, but not routing to any other source/transcoding/passing to voicemail. So I guess my question is, am I correct in the belief that all Asterisk's ISDN channels currently don't support anything other than G.711? How easy would it be to extend one of the ISDN channels to support G.722? Your belief is correct. Right now, the ISDN channels (at least in chan_zap) G.711 is the only voice codec that is supported. I'm not sure what is going to be necessary to get G.722 working there. If it's as simple as changing the bearer capability, the chan_zap work on top of that should be fairly easy. If you have to implement any of the H.* specs to get it working, that will be a bit more trouble. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI debugging ...
Gordon Henderson wrote: On Fri, 16 May 2008, Gordon Henderson wrote: Have a problem with an ISDN30 line in the UK. So following up my own post.. I've not solved this issue, but I think I know what causes it. This was my experiment to put 2 cards in one 1.3GHz system - a TDM400 with 2 x FXO and 2 x FSX and a TE120P - E1 card. The PRI card loses interrupts, so I'm guessing it loses a frame of data when it loses an interrupt, and eventually it gives up and does a reset. The TDM card was rock solid. The system is using oslec too FWIW. When I unloaded the wctdm module the PRI performend flawlessly. So I'm suspecting the 1.3GHz processor and underlying IO is marginal for this application. The Mobo doesn't have an APIC, just old PIC hardware, although both cards were on separate IRQs - the TDM card had the higher priority IRQ though - didn't have time to test it with the cards swapped over, but loading the modules in a differnt order didn't make any difference. Turning off the USB hardware didn't help either. The processor does seem to have a highish high-priority interrupt load (as seen by top). I'll be trying a newer kernel when I get a chance though (this is 2.6.18, compiled to match the motherboard exactly) Making calls through the TDM card just made it worse. However when it was working, it was working very well indeed, but the occasional time when it dropped all calls (about once an hour) wasn't good. You might try turning off echo cancellation to see if your D-channel performance improves. That would be a good test to tell if you should look into perhaps getting either a faster CPU or a hardware echo canceller. It's possible that you may be saturating your poor 1.3 Ghz CPU by doing echo cancellation for too many channels on it. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap channels on TDM400P report to be unimplemented after upgrade from Asterisk 1.2 to 1.4 - (cause 66 - Channel not implemented error)
Steve Totaro wrote: My question is does ANYONE do ANY testing on these releases? It would seem that this bug is so paramount to the purpose of the code that had anyone taken a MINUTE to TEST, it would have been discovered IMMEDIATELY. Not if you already had a zaptel udev rules script installed on the system that's used as the test machine. This was a regression do to recent Makefile changes. A test for this problem has now been added to our pre-release regression testing. Matthew Fredrickson sigh. Thanks, Steve Totaro On Wed, Apr 30, 2008 at 8:41 AM, Steve Totaro [EMAIL PROTECTED] wrote: Sean Bright to Asterisk show details 4:47 PM (15 hours ago) There is a bug in 'make install' in Zaptel 1.4.10 that causes the devices to not be installed correctly. You can either install 1.4.9 or wait for 1.4.11 to be released. On Wed, Apr 30, 2008 at 8:22 AM, Zoran Milenkovic, Datatek d.o.o. [EMAIL PROTECTED] wrote: Hi list! I have RHEL Server rel5 (Tikanga) 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linux with installed digium packets 1. Asterisk 1.4.19 2. Zaptel 1.4.10 3. Libpri 1.4.3 My Digium hardware is [EMAIL PROTECTED] ~]# zaptel_hardware pci::04:00.0 wctdm+ e159:0001 Wildcard TDM400P REV I ...and I have 2 FXO and 2 FXS ports installed inside the TDM400P card The problem is the asterisk doesn't recognize the Zap channels at all. The error is No channel type registered for 'Zap' and Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) and there is the original output form Astersik console: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/zoran-09f1bf90, Zap/3|20) in new stack [Apr 30 13:17:48] WARNING[2845]: channel.c:3001 ast_request: No channel type registered for 'Zap' [Apr 30 13:17:48] WARNING[2845]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/zoran-09f1bf90, ) in new stack == Spawn extension (local, 12, 2) exited non-zero on 'SIP/zoran-09f1bf90' And everything was working quite fine when I was on asterisk 1.2.13, previously installed on this very same server, same Digium card etc. The configurations are totaly the same, also. What could be the resolution of this problem? ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 Released
Matt Watson wrote: Does anybody know if this version fixes the soft lockup during ztcfg using a TE200B? http://bugs.digium.com/print_bug_page.php?bug_id=12468 No, continue to use the stackcleanup branch. That is going to be merged in for the next major release (1.4.11). Matthew Fredrickson -- Matt From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Asterisk Development Team [EMAIL PROTECTED] Sent: Thursday, May 01, 2008 1:07 PM Subject: [asterisk-users] Zaptel 1.4.10.1 Released The Asterisk.org development team has announced the release of Zaptel version 1.4.10.1. This release is a bug fix release for a regression in which the Zaptel udev rules were not installed correctly, as well as a few minor fixes in the xpp drivers. This release is available as a tarball as well as a patch against the previous release. It is available for download from downloads.digium.com. Thank you for your support! ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
Carles Pina i Estany wrote: Hello, We have an Asterisk server with a TE410P Quad-Span togglable E1/T1/J1 card, 3 SPANs configured and OK and one SPAN unconfigured. In our tests it works fine, but when it has a big laod of calls (say, from 40 to 60) we have quality problems: some calls has the sound cut-off (during the call, voice was not stable) The IRQ card is alone, CPU load was not high, network was fine for sure. This server is receiving the calls from SIP channels and routing to the primaries. It's a HP server, multicore, multiCPU. I'm wondering if someone has had these kind of problems (quality problems, sound cut off) with 40 and 60 calls but not with 2 or 3, using Digium cards. Bit later I will call to Digium but I thought that here there is lot of people with lot of experience with these cards. There are a number of factors that can contribute to this type of problem, but probably the best solution is to call support and talk to them about this. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quality problems with ISDN PRI
linuxian iandsd wrote: i have HEARD asterisk wasn't made with the idea to run on multi-core processors in mind .. the result was that it uses one core all the time ..so one single P4 3.4 GHZ would perform better than a far more newser quad one. but i might be wrong. but one thing for sure check hardware compatibility before you buy anything. For the purposes of making sure list records are accurate, this in not true. Asterisk was indeed written with the intention to run on multi-core systems, and should utilize extra cores just fine. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 10:12 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. Ok. So, since the bug we logged was closed and these tests weren't registered along with it, when can one expect to have your new code available in a zaptel release ? In the next one or maybe later because the branch you're working on has lots of different things to merge ? It should be in the next release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: Ex Vito wrote: Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Ok. So here is our reasoning according to collected info. Please correct us where appropriate: 1. The system is supposed to hang while the firmware loads into the DSPs under any zaptel version 2. zaptel 1.4.10 leads to a soft hangup detected, zaptel 1.4.9.2 does not (assuming softhangup detection active in kernel) 3. zaptel 1.4.10 takes much longer ztcfg'ing than 1.4.9.2, that's why the soft hangup is detected under zaptel 1.4.10 (difficult to time, but let's say 1.4.10 takes 10s, 1.4.9.2 takes 3s) Now, back to the original question: - Should this be considered a regression ? - Next steps: a) file a bug and move this analysis to the bug tracker b) don't file bug and move analysis to the dev list c) don't file bug, keep on working on the users list I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. I just realized where this is coming from. I was attempting to patch this from a different angle, but as soon as you mentioned the drastic difference in load time I realized what had happened. I'm going to make another update to my stack reduction branch to see if I can fix this. I'll let you know when it's done. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just realized where this is coming from. I was attempting to patch this from a different angle, but as soon as you mentioned the drastic difference in load time I realized what had happened. I'm going to make another update to my stack reduction branch to see if I can fix this. I'll let you know when it's done. Great. We'll be right here... Since the bug has been closed, we post the timing results we did within this context. I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote: On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: I just updated the branch. Wait about 5-10 minutes in case for the changes to get mirrored, and then try updating and doing it again. Looks better, no more soft lockup and ztcfg time is comparable to 1.4.9.2's: Matthew, ...is there any specific test you'd like us to perform on this revision ? (considering that currently we have no PSTN line to attach to... we can cross-connect the spans and generate traffic or, cross-connect with another lab system) Not really from me specifically. You already tested what I wanted to be tested, and that was to see if I could fix the load time issue and softlockup warning. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
around for host bridges that generate fast back to back transactions which the current version of the quad span cards do not advertise support for. 2008-03-14 16:39 + [r3983-3990] Matthew Fredrickson [EMAIL PROTECTED] * firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 * kernel/wct4xxp/base.c: When doing the ISR rewrite, forgot to include the vpmdtmfcheck when doing DTMF polling causing it to check for DTMF events even when it was told not to (+others) I need to have this system running in about a week and a half. What do you guys say ? The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 3:26 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: The softlockup indicator should be benign. It gets called when loaded the firmware for the part since the firmware image is so large and it takes a long time to load. However, I might have a fix for you. Can you try my stack reduction branch at: https://origsvn.digium.com/svn/zaptel/team/mattf/zaptel-1.4-stackcleanup If that does not work, please contact me directly and I will work with you to get a resolution. Matt, Thanks for your feedback. We've already tested the following branch as per Shaun's suggestion, without getting a different behaviour (see today's earlier email to the list): http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Question: - The url you suggest is very similar, are we talking about a different stackcleanup branch ? We are now in the middle of rebuilding a non 4K stack page kernel so as to give it a try with 1.4.10, the branch Shaun suggested, 1.4.9.2 and the branch you mention, if it is in fact different from Shaun's. We wait your confirmation and will post non 4K stack kernel results later today. One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Shaun Ruffell wrote: Hi Al, Al Baker wrote: Shaun - Could you clarify your post a bit ? 1 - Is the 4 K stacks a Known Problem ? a) If so is it known to be problem on any specific Linux distro ? b) Should ALL installation Check for this PRIOR to doing an Asterisk Install ? I wouldn't really say a known *problem*, since it really depends on what other code is running in the system at the time. I just mentioned that because I've seen 8K stacks help in certain situations. 8K stacks are still the default configuration option in the vanilla kernel. Some distributions (CentOS / Fedora) have switched to 4K by default because they help with memory consumption in highly threaded environments like web servers. For the most part, kernel panics and oops are best handled on a case by case basis with Digium's tech support department since each case is unique. In this case, it looks like his kernel is compiled with the softlockup detector code and it is falsely triggering. Disabling that should remove the warning message at the very least. 2) The branch you mention below - are fixes from it in Any current * release ? They will be in the next Zaptel release. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 4:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing also I would like to see is your kernel .config file. Another thing that would for sure remove that warning is to disable the kernel softlockup detector which is giving a false lockup warning in this case. I belive it's under the KERNEL HACKING configuration menu if you are using menuconfig. Up till now we're running stock CentOS kernel: 2.6.18-53.1.14.el5 The .config is publicly available but we can fwd it to you should you prefer. The kernel we're now building (it is taking quite a while... but it also has been quite a few years since we've built custom kernels... since the 2.0.3x days ?) is based on the stock CentOS kernel with only the 4K stacks option disabled. Please confirm if the SVN branch you suggested is the same or different from the one Shaun suggested yesterday which we already tested. It's the same. Sorry, I sent you that email before I saw his message. I just got an idea for a clever way to make the softlockup detector not complain. I'll let you know when I have a patch to try. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: update with no 4K stack kernel: - The kernel was build from stock centos 5 kernel 2.6.18-53.1.14.el5 - The only .config change was to disable the CONFIG_4KSTACKS Tested zaptel-1.4.10, 1.4.9.2 and the stackcleanup svn branch as suggested by Shaun and Mathew. Short: Results are about the same (stack traces are different). 1.4.10 and the stackcleanup lead to soft hangups, 1.4.9.2 does not. 1.4.10 dmesg snippet: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Matthew Fredrickson Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12xp: Setting up global serial parameters for T1 wcte12xp: Found a Wildcard TE122 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 162 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3613a400 Reg 1: 0x3613a000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff0031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12xp: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 2: Primary Sync Source VPM400: Not Present wcte12xp: Setting yellow alarm VPM450: echo cancellation for 64 channels wcte12xp: Clearing yellow alarm BUG: soft lockup detected on CPU#1! [c044d480] softlockup_tick+0x96/0xa4 [c042de00] update_process_times+0x39/0x5c [c04196ef] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [c0605c30] _spin_unlock_irqrestore+0x8/0x9 [f8e82d57] Oct6100UserDriverWriteBurstApi+0x1d/0x27 [wct4xxp] [f8e95de0] Oct6100ApiLoadImage+0x1b5/0x289 [wct4xxp] [f8e9afc4] Oct6100ChipOpen+0x166/0x25e [wct4xxp] [f8e83050] init_vpm450m+0x196/0x306 [wct4xxp] [f8e6ab11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f8e6eee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042624e] release_console_sem+0x1b0/0x1b8 [c042680e] printk+0x18/0x8e [f8af6fe4] t1_configure_t1+0xc10/0xc18 [wcte12xp] [f8ac65ef] zt_rbs_sethook+0x102/0x13b [zaptel] [f8acdf6a] zt_ioctl+0x273/0x144f [zaptel] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c0483cb3] __d_lookup+0x98/0xdb [c047b32c] do_lookup+0x53/0x166 [c047d9ec] do_path_lookup+0x20e/0x25e [c0471053] get_empty_filp+0x99/0x15e [c047b5a5] permission+0xa2/0xb5 [c04e1a36] kobject_get+0xf/0x13 [c046ea1e] __dentry_open+0xea/0x1ab [c046eb43] nameidata_to_filp+0x19/0x28 [c046eb7d] do_filp_open+0x2b/0x31 [c047f4a7] do_ioctl+0x47/0x5d [c047f707] vfs_ioctl+0x24a/0x25c [c0470de6] __fput+0x13f/0x167 [c047f761] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 3: Secondary Sync Source Completed startup! 1.4.9.2 dmesg snippet: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.9.2 Zaptel Echo Canceller: MG2 PCI: Enabling device :12:01.0 (0150 - 0153) ACPI: PCI Interrupt :12:01.0[A] - GSI 25 (level, low) - IRQ 154 wcte12x[p]: Setting up global serial parameters for T1 wcte12x[p]: Found a Wildcard TE122 Found TE2XXP at base address fdff, remapped to f893e000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x3571b400 Reg 1: 0x3571b000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd0001 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! Registered tone zone 25 (Portugal) wcte12x[p]: Span configured for ESF/B8ZS About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 SPAN 2: Primary Sync Source timing source auto card 0! VPM400: Not Present VPM450: echo cancellation for 64 channels VPM450: hardware DTMF disabled
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito wrote: On Wed, Apr 16, 2008 at 6:51 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote: One thing you can also do is pass the nosoftlockup kernel parameter into the kernel from the bootloader. That should disable the softlockup detector. Tested with no 4K stack kernel and stackcleanup svn branch zaptel version. Correct, the kernel no longer complains about the soft hangup. However the system still hangs (console inoperative, etc) while ztcfg'ing... That is normal while the firmware is loading. It should go away after the firmware has loaded. Can you answer my previous questions ? - If going live would you recommend zaptel 1.4.9.2 or 1.4.10 ? I recommend 1.4.10 by default. However, from what you said it would appear that you are having problems with 1.4.10 so you might stay with 1.4.10 if you are not having any issues with it. - Does the current behaviour from 1.4.10 prevent firmware uploading ? No. There is nothing that is happening that prevents firmware uploading. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem TDM01B
troxlinux wrote: hI list, I have some problems with a TDM01B , when I am talking on the phone with another person it cuts himself the call, this alone I am presented when I make calls to the pstn, with internal extensions I don't have problems I show them the log in the CLI -- Nobody picked up in 68000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/113-081cf588, ) in new stack == Spawn extension (cyber, 2768000, 2) exited non-zero on 'SIP/113-081cf588' Some person of the list that has presented the same problem with this card, and it finds it solved Please contact technical support. You need to get the new version of the firmware for that card, and they will be able to give it to you. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem TDM01B
troxlinux wrote: hI list, I have some problems with a TDM01B , when I am talking on the phone with another person it cuts himself the call, this alone I am presented when I make calls to the pstn, with internal extensions I don't have problems I show them the log in the CLI -- Nobody picked up in 68000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/113-081cf588, ) in new stack == Spawn extension (cyber, 2768000, 2) exited non-zero on 'SIP/113-081cf588' Some person of the list that has presented the same problem with this card, and it finds it solved Sorry, I may have misinterpreted what hardware you have. If you have the new TDM410 card with a hardware echo cancellation module on it, you can get help with a problem similar to that with the new version of the firmware from technical support. If that is not the board that you have, you may have some other issue that you are dealing with. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm410p w/ echo - no full duplex
Michael J. Liberatore wrote: Matthew, I have just emailed support. Do you know what the latest revision is? Also, is it ok for mg2 to be in zconfig.h and echocancel=yes ? It will Yes. Chan_zap and zaptel know how to automatically use the hardware echo canceller. The configuration options like echocancel and echocancelwhenbridged apply the same to hardware and software echo cancellers. Matthew Fredrickson Digium, Inc. know automatically to use the hw ec rather than the software one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, April 11, 2008 11:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm410p w/ echo - no full duplex Michael J. Liberatore wrote: hi, i just installed 2 new tdm410p's on asterisk 1.4.19 with zaptel 1.4.10. They have the hardware echo cancellers. I am having an issue though, when i talk, it cuts out the other end. So for example, i called up another asterisk box and was listening to the prompts and as they were playing if i said something, it would cut out the other end. so i basically started counting and for the 20 seconds i counted, nothing came through from the otherside. i tried from multiple phones and this didnt happen with the old tdm400. is this an issue with the card? Is it because zaptel has mg2 on? Does than mean i am using 2 echo cancellers? the hardware one and the mg2? how should this be set? also, it says echo canceller could not be trained or something like that at the start of every call on the cli. It sounds like you need the new revision of the firmware. Please contact technical support and they should be able to get it to you. Matthew Fredrickson thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. -- -- ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Faraz R. Khan wrote: The newer zaptel (1.4.10) says it includes firmware 1.16 from the CHANGELOG: firmware/Makefile, kernel/wctdm24xxp/base.c, kernel/wctdm24xxp/GpakApi.c, kernel/wctdm24xxp/GpakApi.h: Update wctdm24xxp's VPMADT032 firmware to version 1.16 However there seems to be no way to get this firmware and it does not seem to be included. It checks my firmware and says 1.07 is okay. We had to back that version of the firmware out due to release related problems. As for all problems related to the VPMADT032, if you have any issues, please contact technical support. They will be able to help you with whatever issue you may have. Matthew Fredrickson The URL provided does not contain firmware for the VPMADT032 I* have logged a query with digum. Is there a URL to get this firmware from? On Tue, 2008-04-08 at 11:12 -0500, Ruben Zamora wrote: Lex Thanks, I all ready download the last svn branches from zaptel And i am going to test these afternoon. My phone number es 81-83481611. Thanks Ruben Lex Lethol escribió: Ruben, I am also in Monterrey and have used digium hardware on R2 and PRI. MFC/R2 is not supported by digium but the zaptel driver requirement is the same.. what changes is using libpri vs unicall. Just go ahead and ask them for the firmware update or as Tzafir says use a newer zaptel that should include the updated firmware. If in trouble add me to gtalk I'll try to help out any way possible, Lex On Tue, Apr 8, 2008 at 1:52 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 09:22:30PM -0500, Ruben Zamora wrote: Lex Thanks a lot. These morning i call Digium Support. One issue that i miss in my before e-mail is that i have my Asterisk installed in Monterrey,Mexico and my TELCO provider gave my MFC/R2. Asterisk 1.4.18,Zaptel 1.4.9.2 and Unicall. They told me they can help me because they dont have UNICALL support. So... I need to investigate more or wait for a new zaptel or anything else. Generally you can always use a newer zaptel. -- 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 ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Half-duplex call on TDM2400p with VPMADT032
Ruben Zamora wrote: Hi, I have a same problem, last week i was working with TE120 with a little echo in some call, I replace the card with a TE122B ( Included Echo Cancelation VPMADT032) and there was no more echo in my call. But know i have de same probelm with my incoming audio stream gets clipped / dropped when you speak. Please contact Digium technical support about this. This is definitely something that we need to work with the vendor of the echo canceller IP about. Matthew Fredrickson Thanks Ruben Lex Lethol escribió: Hi, I've used all kinds of digium cards without troubles. My last installation is using a TDM2400p with VPMADT032 echo cancel module and after a week of use we noticed that any incoming audio stream gets clipped / dropped when you speak or when ambient noise is high. The call basically feels as in a half-duplex channel, but only to the person behind our asterisk. I found a quick way to recreate by placing a call using zapata channel, someplace that has an audio stream (ie. music on hold from another pbx). When one talks into the phone, one can notice the incoming audio getting muted until you stop talking. First I thought it had to do with polycom configuration although we use the same setup for all installations (VAD, etc), but the same happens with other sip phones and after more tests I can only recreate this using the TDM2400p's FXO trunks. I have an older TDM2400p with no VPMADT032 in production (without this problem), this leads me to believe there maybe something wrong with VPMADT032 module or with my card in particular. Today I rebuilt everything from scratch using latest asterisk 1.2 release, rechecked with the TDM2400p manual zapata configs just to make sure I wasn't missing something. As the manual suggests, I am just using echocancel=yes and this should set 128 default value for the card. In the general zapata options there we have echocancelwhenbridged=yes. I have played with all yes/no combinations without luck. Interrupts and timing stuff are OK, we have good incoming and outgoing audio quality (as long as its not at the same time). Anyone else using this card showing the same problems? Any zaptel/asterisk gurus wanna take a shot at this? Thanks in advance for your feedback/comments. Lex ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem about voice when using TDM2400p with VPMADT032 echo canceller module
Vu AnhTuan wrote: hi you, I'm having problem with voice quality on my trixbox using TDM2400B.The trixbox is connected via 20 FXO ports on a TDM2400 with the hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a 'background crackle/buzz' coming back when they talk. anyone have the same problem? pls help me. thanks a lot. my trixbox and config file: trixbox version 2.4 (Linux kernel 2.6.18, Zaptel 1.4.7) This is definitely a technical support issue. Please contact them about this so that we can help you get it resolved as soon as possible :-) ! Matthew Fredrickson Digium, Inc. zaptel.conf # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Span 1: WCTDM/0 Wildcard TDM2400P Board 1 fxsks=1 fxsks=2 fxsks=3 fxsks=4 fxsks=5 fxsks=6 fxsks=7 fxsks=8 fxsks=9 fxsks=10 fxsks=11 fxsks=12 fxsks=13 fxsks=14 fxsks=15 fxsks=16 fxsks=17 fxsks=18 fxsks=19 fxsks=20 # channel 21, WCTDM, no module. # channel 22, WCTDM, no module. # channel 23, WCTDM, no module. # channel 24, WCTDM, no module. # Global data loadzone = us defaultzone = us zapata.conf -- ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;default ;echotraining=800 ;default rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes busycount=0 relaxdtmf=yes ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-channels.conf group=1 ;Include AMP configs #include zapata_additional.conf zapata_additional.conf --- ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; ; Span 1: WCTDM/0 Wildcard TDM2400P Board 1 ;;; line=1 WCTDM/0/0 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 1 context=default ;;; line=2 WCTDM/0/1 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 2 context=default ;;; line=3 WCTDM/0/2 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 3 context=default ;;; line=4 WCTDM/0/3 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 4 context=default ;;; line=5 WCTDM/0/4 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 5 context=default ;;; line=6 WCTDM/0/5 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 6 context=default ;;; line=7 WCTDM/0/6 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 7 context=default ;;; line=8 WCTDM/0/7 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 8 context=default ;;; line=9 WCTDM/0/8 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 9 context=default ;;; line=10 WCTDM/0/9 signalling=fxs_ks callerid=asreceived group=0 context=from-zaptel channel = 10 context=default ...more... [IP-PBX ~]# ztcfg -vv -- Zaptel Version: 1.4.7-3259 Echo Canceller: OSLEC Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) Channel 05: FXS Kewlstart (Default) (Slaves: 05) Channel 06: FXS Kewlstart (Default) (Slaves: 06) Channel 07: FXS Kewlstart (Default) (Slaves: 07) Channel 08: FXS Kewlstart (Default) (Slaves: 08) Channel 09: FXS Kewlstart (Default) (Slaves: 09) Channel 10: FXS Kewlstart (Default) (Slaves: 10) Channel 11: FXS Kewlstart (Default) (Slaves: 11) Channel 12: FXS Kewlstart (Default) (Slaves: 12) Channel 13: FXS Kewlstart (Default) (Slaves: 13) Channel 14: FXS Kewlstart (Default) (Slaves: 14) Channel 15: FXS Kewlstart (Default) (Slaves: 15) Channel 16: FXS Kewlstart (Default) (Slaves: 16) Channel 17: FXS Kewlstart (Default) (Slaves: 17) Channel
Re: [asterisk-users] IAXy device
Mojo with Horan Company, LLC wrote: Sean Dennis wrote: bilal ghayyad wrote: Hi All; I have been chocked just when I saw some posts talking about how much the IAXy is bad :) - So I would like to ask, did any one try it later and wether it is good or not? I am asking this because I need to use it as it is NAT Transparent (as I read also, and I did not try it to see how much it is transparent). What about codec? Why it is only support g711 and does not support compressed codec? And what about the IP address and the DNS usage and the DDNS usage? What main porblems contain and any advise? Regards Bilal The device has no echo cancellation and sounds horrible (lots of echo) on about half of the analog phones I tried it on. I wouldn't recommend it unless you absolutely need IAX. It's also very expensive for a 1 port ATA. ___ -- 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 Echo may be the result of latency on the network. I've not had any echo problems that I remember with my IAXy and I make ten calls a day, five days a week, for the last few years, to all sorts of numbers/areas. I know that this isn't representative of typical business use, but residential use, but I've been using in my business and have never been disappointed :) I will agree that's is fairly expensive, but I WOULD recommend it to people who are on the go often. After setup, it really is plug-n-play IMO. Just to put out some official word on the matter, the IAXy does indeed have some echo cancellation built in. It has to since it interacts with a phone via a 2 wire to 4 wire conversion with a hybrid. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call deflection on ISDN PRI in Sweden
Hanna Wallin wrote:
Hello List!
We're having trouble making call deflection on ISDN PRI. We would like to
transfer a call to an external extension but keeping the callerid of the
caller so it can be presented to the receiver of the transferred call.
At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
TE420B. We've ordered the service (CD) from the phone company.
The zapata.conf file inlcludes:
Transfer= yes
Facilityenable=yes
Callerid=asreceived
In extensions.conf we try to transfer a call to an external extension as:
Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
UNSUPPORTED.
Ideas anyone? We would really appreciate it!
That supplementary service (CD) is not supported in libpri right now, so
that would be the reason why it doesn't work. The Transfer()
application is for analog lines, IIRC.
--
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.
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Re: [asterisk-users] Problem: Digium TDM400 with XOptionsFlex - Solved
Thomas Klettke wrote: On Sat, 2008-03-22 at 12:48 -0400, John Novack wrote: Assuming you have also checked the obvious possible defects regarding cords from the XO device to the Digium card, what happens if you reverse tip and ring? John, you were right on the money: I've found that the two lines that gave me problems had the polarity reversed. Correcting it solved the problem. I wish I had checked that last week - before spending hours on troubleshooting ... Not certain even if the Digium FXO circuit is even sensitive to line polarity, Apparently it is - unlike the Sangoma A200 which worked with either polarity. Thanks for your help I can't say how much I appreciate it. Let me know if you're ever in the Houston area: I'll buy you a beer, or two ;-) Cheers, Thomas John Novack Just to let you guys know, we're looking into this to see why this might be happening. We'll keep you posted when we find out what's wrong. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Issues
Jeremy Mann wrote: Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the lock-up, IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is asterisk as a man-in-the-middle, Span 1 goes to Telco, Span 2 to Nortel MICS. PRI is not the problem as it's plugged into the Nortel directly for now and we have no problems. Nothing in dmesg indicates any errors. Any clue how I go about debugging this? The best way is to start going through versions and figuring out which version it broke at. Some other things worth checking: What versions of Zaptel/libpri/Asterisk did you upgrade from? When you upgraded, did you recompile them in the correct order (Zaptel 1st, then libpri, then Asterisk)? Matthew Fredrickson [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 57 from conference 9/1 [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Unlinking slave 26 from 3 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 36 from conference 9/3 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 14 from conference 9/26 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/26-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/26-1 [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/3-1 [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/21 [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/20 [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Found empty available channel 0/19 [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann el. [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Found empty available channel 0/18 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXOTUNE update
Hey all, First of all, some background: Fxotune is a utility that is used to tune the hybrid on FXO modules For all of you with FXO modules out there, fxotune can help you adjust the analog and digital hybrid that is on the FXO interface and tune it so that it maximizes echo return loss. This means that it will reduce your default echo which is received, and will help any echo cancellers on the line to do a better job. If using one of the open source software echo cancellers, using fxotune can be the difference between having echo problems and not having echo problems. This is the update: I just committed a new version of fxotune which uses a better technique for measuring echo return loss. Before, there was a simple power calculation which was done on the samples that would indiscriminately check the power of all samples received. This works well when the line is silent, but if there are any sort of tones in the background or noise due to noisy line conditions, this calculation can yield results which may improve things, but are not the best results. The new method involves using fourier analysis of the tones used in the test reference which is sent out. Using fourier analysis instead of the power calculation, we can cut through any background noise which is not related to our test sequence's set of tones, producing a much more accurate and noise immune calculation. If you have run fxotune before on your lines, I recommend you re-run it with the updated version of the utility. As of this moment, it is not yet in a released version of zaptel, but if you check out either latest 1.2 or 1.4 branches, it will be there. If you run fxotune with the -v option, it will tell you what the return loss it calculates for each AC impedance and set of coefficient parameters in dB. In order to use the new analysis calculations, you do not need to pass any sort of special parameters to fxotune, it does the new analysis technique by default. Please let me know if you have any issues as well. Thanks! -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IBM x3400 w/ Digium TE220
Edwin Lam wrote: hi folks. i have a Digium TE220 PCI-E 2 port T1/E1 controller installed in an IBM x3400 server. i load the wct4xxp driver seems ok. but when i execute ztcfg -vvv command. the kernel panic. i tried zaptel 1.2.21 22. they have the same result. following is my zaptel.conf: loadzone=cn defaultzone=cn span=1,1,0,ccs,hdb3 span=2,0,0,esf,b8zs bchan=1-15 dchan=16 bchan=17-31 fxoks=32-55 any clues? p.s. the same setting works fine on HP Proliant server. This looks like a really good reason to call Digium tech support :-) It's comes free with the purchase of the card. I haven't heard of anything like this, although posting your kernel panic output would help. But it would be best to handle this through tech support. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Increasing the voice volume from the diguim cards
bilal ghayyad wrote: Hi List; Anyone knows a method (command) to increase the voice volume at diguim card level? Are you trying to do this at some other level than rxgain and txgain settings in zapata.conf? If so, for the analog cards there are some module parameters for doing so. For digital T1/E1 cards, the only way to do it is with the gain options in zapata.conf. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
Daryl G. Jurbala wrote: How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and asterisk would stop accepting IAX connections in less than a day and would need to be restarted. It has been a continuously worked on task (ever since a few months ago). Russell Bryant and others have been working on it and has improved its reliability to the point of fixing most if not all of the previously outstanding issues. I recommend trying it again. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TE120P versus Sangoma A101D-X
Paul Hales wrote: I also understand your stand here Kevin - there is no way you can restrict the software running on a server out in the wild, and no way to make sure the software they are running will not conflict in any way. But a single port E1 card with hardware echo cancellationpossible? Hold that thought just for a little bit :-) -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem dialing certain numbers with an E1 PRI
Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T309 Timer: -1 T313 Timer: 4000 N200 Counter: 3 ___ --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 -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI
Jacob Lefkowitz wrote: I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error:OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error:ERROR: RLT Not Allowed I have tried on two different DMS500 switches with two different phone companies. The phone companies swear it is enabled on their end, and they are billing accordingly :). This is using Asterisk 1.4.13/Zaptel 1.4.5.1 (although it also did the same on earlier 1.4 versions). Has anyone been successful with this? Yes, I personally saw it work successfully (I wrote the code), and know it has been deployed on many systems. Maybe you should ask the other end why the other end is saying RLT not allowed? Also, you have to make sure that it is between an inbound call (to Asterisk, from DMS) and an outbound call (from Asterisk to DMS). It should already check for this in libpri, but I figured I'd mention it just to be sure. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sangoma zaptel patches
Steve Totaro wrote: Dovid B wrote: - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 11, 2007 8:21 PM Subject: Re: [asterisk-users] sangoma zaptel patches On Sunday 11 November 2007 11:07:04 Steve Totaro wrote: Tzafrir Cohen wrote: Sangoma's s setup process includes a small patch to Zaptel. I have some technical reservations with that patch. It seems that under certain circumstances it may cause unexpected behavior when used with other Zaptel channel drivers. I also don't understand why a safer method is not used. Just out of curiosity, I have yet to see any issues with Sangoma cards and the way they ride on top (and patch) the Zaptel drivers. This personal dataset is around one hundred productions boxes. How many of those boxes are of the type that Tzafrir is worried about? Specifically, how many of those boxes contain a combination of telephony hardware from vendors other than Sangoma? I have a box that now has a TDM400P. I will be installing a sangoma card in it soon and I actually need support for this. I set up almost the exact same configuration and all went well (HP DL380). No gotchas or glitches. I have a feeling that Tzafrir is trying to fix what is not broken, since he never pointed out a single conflict between various hardware using patched Zaptel drivers configurations. Maybe he is looking down the road and being proactive which I applaud, but I think he is obsessing over what he feels is the incorrect way of doing things and demanding (tone in emails) that they cooperate and do what he tells them. A little tact goes a long way. I think that part of it is that the patch that they do to zaptel replicates existing zaptel functionality (zt_hdlc functions) for hardware d-channel support. There has been no change in their patch to use these existing functions, and they are implementing this via an ioctl function within a kernel driver, which is not a pretty way to do what they are trying to do. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please explain the correct LED color for B410P
[EMAIL PROTECTED] wrote: Hi. I have installed B410P in Europe and the cards works more or less ok. My question is what color should the LED's on the back of the card be when connected to the PSTN NT box? Is there anywhere some information on the expected LED color in any given state (idle, call active, cord unplugged etc.)? On my card the lights are shining Red(orange-ish) but flashing to green every now and then and then shining green when there is a call on one of the lines for that port. That is correct. On zaptel-1.4/misdn-1.1.x you should see a blinking red when layer 2 is not up, constant red when layer 2 is up, and it will flash green when D-channel messages are sent on the port. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two PRI setup questions
Tilghman Lesher wrote: On Thursday 01 November 2007 19:31:39 Lutgring, Sam wrote: 2) Is there a way to see the idle status of a B channel? When ATT tells me they don't see the B channels coming up, is there a way that I can see this in Asterisk??? Ask ATT to turn off B channel maintenance protocol on the PRI. Asterisk does not support this mode. It may work with either the 4E or 5E switchtypes. There is some code that (I didn't write it) I think unbusies the channels that is executed for one or the other switchtypes. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
