[asterisk-users] Info 16.13.0 with SNOM FW 8.7.5.35
Hello, I found a problem with latest Asterisk 16.13.0 and SNOM3xx phones (EOL) running the above FW. Incoming calls are no more working, we get error 404 despite the fact that broken registrar is on for the account. Previous FW for this phones don't have the problem. At this time I open a ticket at SNOM support, don't know if I should also open an issue at jira. Regards -- Daniel -- _ -- 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] Info 16.13.0 with SNOM FW 8.7.5.35
On Wed, Sep 9, 2020 at 7:00 AM Administrator wrote: > Hello, > > I found a problem with latest Asterisk 16.13.0 and SNOM3xx phones (EOL) > running the above FW. Incoming calls are no more working, we get error > 404 despite the fact that broken registrar is on for the account. > Previous FW for this phones don't have the problem. > > At this time I open a ticket at SNOM support, don't know if I should > also open an issue at jira. > If you upgraded the phone and it stopped working, then the issue would be with the phone so an issue should be opened with the vendor. If you upgraded Asterisk and it broke the phone then the issue would be with Asterisk and an issue should be opened on JIRA. -- 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
[asterisk-users] INFO to rfc2833 issues
Hi, We have a box running Asterisk 11. A call comes in and the caller wants to use INFO (and the peer is set as INFO). We send the call out to a carrier were we specify rfc2833 and negotiate it correctly. In theory Asterisk should see the DTMF in rfc2833 and convert it over to INFO when the called person presses a button but this does not happen. Asterisk simply sends along the rfc2833 packets. Is there something that I am missing here (other then upgrading asterisk)? TIA. Dovid -- _ -- 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] info about DID.
On Thursday 27 Oct 2016, KyD wrote:
> Hi!
>
> I need to make a dialplan by DID.
>
> where it gets the asterisk values did? from sip headers or ... ?
>
> Thanks!
It will all be taken care of for you, so you don't have to do anything special
for calls to a direct inbound number. When a call comes into your Asterisk
via a SIP trunk or ISDN line and hits the appropriate context in your
dialplan, the destination number (i.e., what the person on the other end
dialled) will be in the variable ${EXTEN}. (But *note*, it may be in any one
of several formats: local number only; national number with STD [town]
code, with or without initial 0; international number with IDD [country] and
STD [town] codes). The same provider will always present the number in the
same format, though; so if they include the STD code, they will -always-
insert it, even for calls within the same town where the caller dialled only
the local number.
If you define an "s" extension which displays the value of ${EXTEN} in the
console, then dial one of your direct lines, you can work out how the number
is being presented:
exten => s,1,NoOp(Incoming call for ${EXTEN})
Then you can write appropriate extension logic in your dialplan. For
instance, if one of the dialled numbers comes in as "206318", you just need to
have an extension like this in your dialplan;
exten => 206318,1,DoSomething()
And when somebody dials that number, it will fire the appropriate extension in
your dialplan.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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[asterisk-users] info about DID.
Hi! I need to make a dialplan by DID. where it gets the asterisk values did? from sip headers or ... ? Thanks! -- KyD GNU/Linux SysAdmin Quanto mais você sabe, mais você percebe que você não sabe nada. -- _ -- 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] info on : CONFIG_VOICEBUS_ECREFERENCE
Hi, just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE undef in the voicebus then how it will defined to run the echo cancell on the respective drievers wctdm24xxp ?? explain how this CONFIG_VOICEBUS_ECREFERENCE enabled and where it is enabled while run time. regards Uppi. -- _ -- 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] info on : CONFIG_VOICEBUS_ECREFERENCE
On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote: Hi, just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE undef in the voicebus then how it will defined to run the echo cancell on the respective drievers wctdm24xxp ?? explain how this CONFIG_VOICEBUS_ECREFERENCE enabled and where it is enabled while run time. It's a compile time option that was in added to the code for testing in certain environments [1]. It doesn't enable / disable the echocanceler, but instead changes the default processing to provide a more accurate reference signal when latency grows large. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9144 If you would like to use it for some reason, in drivers/dahdi/voicebus.h change #undef CONFIG_VOICEBUS_ECREFERENCE to #define CONFIG_VOICEBUS_ECREFERENCE. In practice, I've found that it's generally better to fix what on the system is causing latency to grow so large instead of enabling that option. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE
Hi , it implies that under this CONFIG_VOICEBUS_ECREFERENCE code inside this condition will never execute. if we need it should be defined . Regards Upendra On Tue, Nov 20, 2012 at 8:55 PM, Shaun Ruffell sruff...@digium.com wrote: On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote: Hi, just going through the code i found that this CONFIG_VOICEBUS_ECREFERENCE undef in the voicebus then how it will defined to run the echo cancell on the respective drievers wctdm24xxp ?? explain how this CONFIG_VOICEBUS_ECREFERENCE enabled and where it is enabled while run time. It's a compile time option that was in added to the code for testing in certain environments [1]. It doesn't enable / disable the echocanceler, but instead changes the default processing to provide a more accurate reference signal when latency grows large. [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=9144 If you would like to use it for some reason, in drivers/dahdi/voicebus.h change #undef CONFIG_VOICEBUS_ECREFERENCE to #define CONFIG_VOICEBUS_ECREFERENCE. In practice, I've found that it's generally better to fix what on the system is causing latency to grow so large instead of enabling that option. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Info on using LDAP with Asterisk?
On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id291590 Thanks! Leif. -- _ -- 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] Info on using LDAP with Asterisk?
Thanks, That's more info than I've been able to find to date. I'll work on digesting it now. On Mon, Jan 24, 2011 at 7:37 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-01-23 02:56 PM, Jeff B wrote: There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? We've tried to document some of it here: http://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id291590 Thanks! Leif. -- _ -- 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] Info on using LDAP with Asterisk?
On Mon, Jan 24, 2011 at 10:22 AM, Jeff B jeffb.l...@gmail.com wrote: Thanks, That's more info than I've been able to find to date. I'll work on digesting it now. Please add you comments and findings to https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver to help us have a good source of information. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Info on using LDAP with Asterisk?
There does not seem to be very much info out there about using LDAP to create asterisk configurations. Does anyone have some information that they would suggest I start with? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] info about application not available asterisk 1.6.2.11
Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -- _ -- 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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens jonas.kell...@telenet.be wrote: asterisk*CLI core show application Dial did you have libxml-doc installed when you build asterisk? *CLI module load app_dial.so -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 11:37 AM, Paul Belanger paul.belan...@polybeacon.com wrote: did you have libxml-doc installed when you build asterisk? s/-doc/-dev -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] info about application not available asterisk 1.6.2.11
On 09/09/2010 05:37 PM, Paul Belanger wrote: On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellensjonas.kell...@telenet.be wrote: asterisk*CLI core show application Dial did you have libxml-doc installed when you build asterisk? *CLI module load app_dial.so Indeed I did not had libxml-doc installed and so I build asterisk 1.6.2.11 without it. I did not know the consequence when installing... Can I install libxml-doc now without having to rebuild asterisk ?! Kind regards, Jonas. -- _ -- 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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Can I install libxml-doc now without having to rebuild asterisk ?! No, install libxml-dev then rerun ./configure, make install -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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] info about application not available asterisk1.6.2.11
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Subject: Re: [asterisk-users] info about application not available asterisk1.6.2.11 On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Can I install libxml-doc now without having to rebuild asterisk ?! No, install libxml-dev then rerun ./configure, make install -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com The reason for this is that the configuration in place is not aware of libxml, so just re-making Asterisk without configuring will result in a continuation of the existing problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] info for Busy for incoming internal call but not for exterrnal
Hi all, I've an asterisk ver 1.4.22. As in object I have an extension beloning to a queue. I need that for an external incoming call, the extension recieve the call waiting signal/tone, whereas for internal incoming call, the extension appear busy. Is it possible? Could someone let me know the right way to do that? thanks a lot in advance lorenzo -- Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non conta un cazzo, 1941 ... sono anche un autore -- _ -- Bandwidth and Colocation Provided 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] Info about dstchannel
Hi,
Is it possible to get information about SIP destination channel (created
after Dial command) somehow?
For example I would like to know what codec was used. I can do this for
originating channel with:
${CHANNEL(audionativeformat)}
but not sure how to do the same for destination channel?
Any suggestions?
Chris
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Re: [asterisk-users] Info about Providers
Al, If you don't mind I am actually writing to you about a little different matter. I am new to the asterisk biz and am not too far from where you are located in PA. I am interested in starting a VOIP business like your own and was wondering how you are finding the market for new customers? I noticed that you provide many services on your site. I am interested in providing call center services and am starting to see if there is a market for this still. I would like to know your thoughts since you are already out there in the trenches. Thanks, Al On 7/13/07, Al Bochter [EMAIL PROTECTED] wrote: To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fairly open and truthful And this list will be carefully moderated Please do some posting! By the way I am looking for moderators for the list if you want to help let me know. -- Best regards, Al Bochterhttp://www.BochterServices.com http://www.bochterservices.com/ --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online storehttp://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Info about Providers
To everyone on the list I put a site on line the URL is *http://bochterservices.com/phpbb/ *This is for any information on Good or Bad ITSP You can post any problems you had with the provider You can Vote on the provider This is for allowing multiple viewpoints to be heard. If a provider receives a bad review, they are more than welcome to post So long as the exchange is fairly open and truthful And this list will be carefully moderated Please do some posting! By the way I am looking for moderators for the list if you want to help let me know. -- Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- ___ --Bandwidth and Colocation Provided 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] Info: Nokia E65 working with Asterisk
Hi, Just for information on compatibility: Earlier this week I got a Nokia E65 which supports WiFi and SIP. I got the WiFi side configured to work with an access point after several attempts. This eventually had to be done using all manual settings, as using it's config wizard gave WEP Key errors despite many attempts and careful verification. It's WiFi system is not particularly sensitive, it does not detect an AP at the far side of the building that is quite useable from the this location by a typical notebook PC. This may be an advantage, as it's not going to connect to anything with a weak signal that could drop out with movement. I set up a standard SIP extension for it in asterisk via freepbx. Some of the phone's SIP settings are less than obvious, at least to me (i.e. where to put the Asterisk box's IP), but I found an excellent guide here: http://newlc.com/Using-SIP-with-Nokia-Series60-and.html After adjusting the settings in line with that, it works perfectly. I've also set to be permanently registered so it's available for incoming calls while in range. I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Voice quality is excellent and I've not had any problems with it so far. Robert Jenkins. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Info: Nokia E65 working with Asterisk
I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Is this 'Call' go to 'Internet Call' usable when you select a callee using the phone's directory ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Info: Nokia E65 working with Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 8 Mar 2007, at 13:34, Olivier wrote: I have left the default for outgoing calls to be the mobile network. To make a call via the Asterisk PBX, you need to enter the number then press the 'options' key, select 'Call' go to 'Internet Call'. Is this 'Call' go to 'Internet Call' usable when you select a callee using the phone's directory ? Yes it is. However, this also depends on how you set up your dial plan and how you store phone numbers in your directory. I have set up my Asterisk dial plan to understand and work with the universal phone number notation of +country codearea codenumber, which is understood by the mobile network as well. I store all my phone numbers that way, be they local, long distance or international long distance from where I am. This means I can select any phone number from my phone book and dial out via the mobile network or my Asterisk server, it just works. jens -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (Darwin) iD8DBQFF8AnFRAx5nvEhZLIRAqPbAKCH2IxZAvTTtt4D8WjbzU5WVz6FGACfTVD6 bAaLd67dNaiatajZ3nSdP4A= =V36x -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO: TFOT book- n priorities and labels
Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled Unnumbered Priorities. http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] INFO: TFOT book- n priorities and labels
On Wed, 2006-05-31 at 02:01 -0700, Michael Collins wrote: Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled Unnumbered Priorities. http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip And one of the authors of that book (Jim Van Meggelen) will be speaking at ClueCon (on asterisk topics I believe) in august if you want to talk to him in person :) for more info see http://www.cluecon.com/ -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Info
The line build out value is a power level that is set based on the distance from the Device to the T-1 service provider's gateway. If the device is close by, the gateway requires less power and the line build out value is lower; if the device is far away, the gateway requires more power and the line build out value is higher. This setting is determined by the T-1 service provider. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Friday, May 05, 2006 7:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Info Hi all, anyone could pls explain me what does it means ? It a part of zaptel.conf file. LBO= Line Build Out 0:0dB(CSU)/0-133feet(DSX-1) 1:133-266feet(DSX-1) 2:266-299feet(DSX-1) 3:399-533feet(DSX-1) 4:533-655feet(DSX-1) 5:-7.5dB(CSU) 6:-15dB(CSU) 7:-22.5dB(CSU) Thanks Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/333 - Release Date: 05/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info
Alexander Lopez wrote: The line build out value is a power level that is set based on the distance from the Device to the T-1 service provider's gateway. If the device is close by, the gateway requires less power and the line build out value is lower; if the device is far away, the gateway requires more power and the line build out value is higher. This setting is determined by the T-1 service provider. The LBO for all T-1s that I've ever seen has been from the smartjack, not the CO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Info
I guess that depends on the type of loop that serves you. If it is a HDSL loop then the SmartJack would be responsible for the build out. If it is a traditional T1 loop it would be set from the CO. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Saturday, May 06, 2006 2:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Info Alexander Lopez wrote: The line build out value is a power level that is set based on the distance from the Device to the T-1 service provider's gateway. If the device is close by, the gateway requires less power and the line build out value is lower; if the device is far away, the gateway requires more power and the line build out value is higher. This setting is determined by the T-1 service provider. The LBO for all T-1s that I've ever seen has been from the smartjack, not the CO. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info
Hi all, anyone could pls explain me what does it means ? It a part of zaptel.conf file. LBO= Line Build Out 0:0dB(CSU)/0-133feet(DSX-1) 1:133-266feet(DSX-1) 2:266-299feet(DSX-1) 3:399-533feet(DSX-1) 4:533-655feet(DSX-1) 5:-7.5dB(CSU) 6:-15dB(CSU) 7:-22.5dB(CSU) Thanks Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.392 / Virus Database: 268.5.5/333 - Release Date: 05/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info system
I would like to set-up a info system. A. weather I found the code to get the weather report and with festival I can create the audio file. However, I have different location of my useres, which are easy to distinguish with the users starting phone number. E.g. users in region A would start their phone number with 54, while region B would start with 73, How can I give the user the right weather report according to their own phone number (without an agi) ? B. Music charts I would like to pull down from www.garageband.com new music and let users play this music. 1. user should be able to set the quality (codec) 2. user should be able to vote for the music 3. is there a possibility to use video as well? C. Gateway ratings I am planning to announce the gateway number to the user. I also would like to get a rating of the last phone call from the user. I am not really sure what I info I want nor how I want this info. This info could be just a rating of the gateway for the last called phone number with: call time, called number, caller, gateway number. More sophisticated would be to pull out all info we get from this call (from the debug file) and send it by email to me. Any ideas? Any starting points? (other than google or wiki) Has somebody experience with such things? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about mp3 which are installed with Asterisk
asterisk tends to not work well with mp3's that have ID3 tags --- Zach A [EMAIL PROTECTED] wrote: Hi, The 3 MP3 files which are installed with asterisk, what is their bit rate, are they mono and do they have ID3 tags? Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info about mp3 which are installed with Asterisk
Hi, The 3 MP3 files which are installed with asterisk, what is their bit rate, are they mono and do they have ID3 tags? Zach A ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about mp3 which are installed with Asterisk
Zach A wrote:
Hi,
The 3 MP3 files which are installed with asterisk, what is their bit
rate, are they mono and do they have ID3 tags?
Zach A
{192}([EMAIL PROTECTED]:Desktop)# file /var/lib/asterisk/mohmp3/*.mp3
/var/lib/asterisk/mohmp3/fpm-calm-river.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-sunshine.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
/var/lib/asterisk/mohmp3/fpm-world-mix.mp3: MPEG ADTS, layer III, v1,
128 kBits, 44.1 kHz, JntStereo
{275}([EMAIL PROTECTED]:Desktop)$ id3tool
/var/lib/asterisk/mohmp3/*.mp3
Filename: /var/lib/asterisk/mohmp3/fpm-calm-river.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-sunshine.mp3
No ID3 Tag
Filename: /var/lib/asterisk/mohmp3/fpm-world-mix.mp3
No ID3 Tag
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level, I'm really quite busy.
___
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Re: [Asterisk-Users] Info about F1000G
Tomislav Parčina wrote: Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. Bought one as a sample. Build quality is above average. Battery life is very short. Maximum volume of the headpiece is very very low, e.g. you can't hear anything while talking on the street or in a noisy environment like a bar, etc. Menu is not user friendly, sip and network settings are too far from each other. Doesn't support wifi networks that use 802.1x billing systems. Resume: OK phone for home / small office use, but you'll always want to get a better one. Has become a member of Hall of Junk for our HQ museum. ;) HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Info about F1000G
Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does allow login to web-portal protected billed wifi systems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vahan Yerkanian Sent: Thursday, March 02, 2006 12:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Info about F1000G Tomislav Parčina wrote: Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. Bought one as a sample. Build quality is above average. Battery life is very short. Maximum volume of the headpiece is very very low, e.g. you can't hear anything while talking on the street or in a noisy environment like a bar, etc. Menu is not user friendly, sip and network settings are too far from each other. Doesn't support wifi networks that use 802.1x billing systems. Resume: OK phone for home / small office use, but you'll always want to get a better one. Has become a member of Hall of Junk for our HQ museum. ;) HTH, Vahan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about F1000G
Hello Tomislav, I borrowed F1000 from my friend for testing. I am not sure if that is different from F1000G, but I am experiencing the following issues: 1. As a user, it is not easy to get a firmware update as I need to have a service contract. 2. Even with the latest firmware I got from sipgate.de (version 3.80st), I can only have WPA-PSK with TKIP encryption, while I prefer AES. 3. The voice quality is sometimes really bad when using codecs with compression (G729 and G726). No problem with G.711. 4. The battery does not last long, just around 22 hours. I don't have any other issues a part from those. Cheers, Anto - Original Message - From: Tomislav Parčina [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 02, 2006 8:45 AM Subject: [Asterisk-Users] Info about F1000G Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info about F1000G
wendell hamilton wrote: Ditto on the volume issue, it's EXTREMELY low, but the latest firmware does allow login to web-portal protected billed wifi systems. You don't want web-portal based auth schemes for your metropolitan wifi network, as someone can use your network as a transport with two wifi adapters located on the opposite ends of your city using CIDR ip addresses and eating available bandwidth. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info about F1000G
Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info request from Sangoma users
Hi :) I have an A104 and wondered if other owners could confirm the strange behaviour I'm seeing.. it's best seen on an idle system, thus eliminating asterisk or other factors.. Very simply, just let 'vmstat 1' run for a few minutes and watch the output, specifically the 'sy' column... On the 2.4G Xeon machine I'm using, the system CPU usage sits very low for a minute or two, and then spikes up to 100 for a few seconds, before tailing off again - this happens all the time :( Interestingly, the 'load average' as reported with 'w' always stays at zero even with this high 'system load'... I moved the card to another PCI slot (and bus) and get the same thing, but now much more frequently, but for a much shorter length of time... Now bringing Asterisk into the picture, I can't use Monitor() because once I get even 5 simultaneous recordings, the real 'load average' on the machine spikes up to 2 and greater, and calls become stuttered as the machine fails to keep up with whatever it's doing.. The machine is SCSI, with a decent LSI Logic onboard controller and fast disks - so it's nothing to do with enabling DMA - hdparm shows 70MB/sec with minimal load increase. Can anyone confirm this behaviour? Cheers, Gavin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
Hello, Can you post what firmware your board is and what wanpipe driver version you are using? We do up to 50 concurrent recordings on our systems and they do not have recording issues. We use MegaRAID 320-1 cards as well. MATT--- On 12/13/05, Gavin Hamill [EMAIL PROTECTED] wrote: Hi :) I have an A104 and wondered if other owners could confirm the strange behaviour I'm seeing.. it's best seen on an idle system, thus eliminating asterisk or other factors.. Very simply, just let 'vmstat 1' run for a few minutes and watch the output, specifically the 'sy' column... On the 2.4G Xeon machine I'm using, the system CPU usage sits very low for a minute or two, and then spikes up to 100 for a few seconds, before tailing off again - this happens all the time :( Interestingly, the 'load average' as reported with 'w' always stays at zero even with this high 'system load'... I moved the card to another PCI slot (and bus) and get the same thing, but now much more frequently, but for a much shorter length of time... Now bringing Asterisk into the picture, I can't use Monitor() because once I get even 5 simultaneous recordings, the real 'load average' on the machine spikes up to 2 and greater, and calls become stuttered as the machine fails to keep up with whatever it's doing.. The machine is SCSI, with a decent LSI Logic onboard controller and fast disks - so it's nothing to do with enabling DMA - hdparm shows 70MB/sec with minimal load increase. Can anyone confirm this behaviour? Cheers, Gavin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 07:24 -0500, Matt Florell wrote: Hello, Can you post what firmware your board is and what wanpipe driver version you are using? Hi Matt :) I've already been through all this with Sangoma's support - just looking for external opinions from real-life installs - so thank you for the response :) I've seen this behaviour with everything from the first 2.3.2 Asterisk-compatible wanpipe to the latest 2.3.3-beta18. We do up to 50 concurrent recordings on our systems and they do not have recording issues. We use MegaRAID 320-1 cards as well. That's what I thought - I mean the amount of disk IO is absolutely nothing at all :( What kind of CPUs are you using? Also, single or dual (or a single with hyperthreading ?) What onboard L2 cache do they have? My last hope is to try a P4 machine with 1MB cache, since the others I've used have 512K.. They're all Dell machines - and I know the reaction that usually evokes when dealing with Digium hardware (been there, seen that...) - I thought someone like Sangoma with many more years in the business would be more immune to things like this :( Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 12:59 +, Gavin Hamill wrote: [snip] What kind of CPUs are you using? Also, single or dual (or a single with hyperthreading ?) What onboard L2 cache do they have? My last hope is to try a P4 machine with 1MB cache, since the others I've used have 512K.. They're all Dell machines - and I know the reaction that usually evokes when dealing with Digium hardware (been there, seen that...) - I thought someone like Sangoma with many more years in the business would be more immune to things like this :( Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? Think you can do it either in the BIOS or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. There is some boot option called nmi_watchdog that can be set at 0 or 1 that perhaps solves that one. When things get really weird try reseating the memory modules. And if you have a dual Xeon box and only one cpu shows up when booting Linux try reseating the processors too. While you are at it reseat everything you can find :) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote: Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :) or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. Yes I had those with the Digium card (before I returned it, obviously :), although Digium support managed to solve those in the driver. While you are at it reseat everything you can find :) Feel the build quality :)) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). All very sage advice - I have another box to try it on yet before curling up in a corner and crying - I'll report back if I find anything spectacularly wrong :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info request from Sangoma users
We use all Asus motherboards now, with single P4 processors(some with 512k, 1024k and 2048k L2 caches) We run most of them with HT on, no issues there. Also, if you are using the on-board RAID, it's not really a complete LSILogic RAID, They(LSILogic) won't support it because Dell does modifications to the hardware and firmware to optimize it's performance. Many calls to Dell and LSILogic left me very frustrated about this. I now personally avoid Dell servers at almost all costs. (I've even refused a free one offered to me) I've just had too many issues with them in the past(and Compaq too). Now we build all of our servers ourselves and can't be hapier about it. And with the money we save we buy replacement parts to keep on hand and have a spare server ready to replace any of our production servers at a moment's notice. Good luck, MATT--- On 12/13/05, Gavin Hamill [EMAIL PROTECTED] wrote: On Tue, 2005-12-13 at 14:32 +0100, Patrick wrote: Been a while since I used Asterisk on a Dell box but I remember I had to turn off HT. Have you tried that? For sure, I've tried HT on and off, with 2.4 and 2.6 kernels :) or booting the kernel with noht. On Dell boxes I have also seen some funky NMI received for unknown reason. Dazed and confused messages in /var/log/messages. Yes I had those with the Digium card (before I returned it, obviously :), although Digium support managed to solve those in the driver. While you are at it reseat everything you can find :) Feel the build quality :)) As a test you can also disable the onboard nic and stick in a quality nic on its own interrupt to see if that helps. And off course disable in the BIOS everything that you do not use (serial/parallel/usb etc.). All very sage advice - I have another box to try it on yet before curling up in a corner and crying - I'll report back if I find anything spectacularly wrong :) Cheers, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info on beta1 seem to be broke
I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends DTMF via sip info packets to another beta1 box. The peer is set to receive info. What I get is a click sound and a very very short tone. Sound like to me that I get the first part of the tone before it is captured and put in the info packet, but the gateway never seems to send the tone, the packet that gets sent looks like this: -- -- SIP read from 192.168.117.4:5060: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.117.4:5060;branch=z9hG4bK7bd677b8 From: WIRELESS CALLER sip:[EMAIL PROTECTED];tag=as37dd610c To: sip:[EMAIL PROTECTED];tag=as3af9dc41 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: ISDN-NET Voip Gateway Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=500 --- I know it is not the receiving box messing things up as I get the same short short DTMF sound on a cisco IAD. Something is wrong with this packet but I just can't see it!!! Is there any rtp that gets sent, anyone know what the Content-length does? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on beta1 seem to be broke
After reading through the rfc http://www.ietf.org/rfc/rfc2976 and cisco site http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t11/ftinfo.htm And googling a bit I can not find anything that look out of place, I notices a few formating difference on examples I have seen and most examples have a content-length: 26 but I don't think that should matter, and the rfc does not look like there is any rtp needed for info application/dtmf-relay packets. Could if be as simple and the space after the = INFO sip:201 at 192.168.1.38 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.36:5060 From: sip:101 at 192.168.1.36;tag=43 To: sip:201 at 192.168.1.38;tag=9753 Call-ID: 100450864100 at 192.168.1.36 CSeq: 3 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2 Duration= 110 James Sizemore wrote: I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends DTMF via sip info packets to another beta1 box. The peer is set to receive info. What I get is a click sound and a very very short tone. Sound like to me that I get the first part of the tone before it is captured and put in the info packet, but the gateway never seems to send the tone, the packet that gets sent looks like this: -- -- SIP read from 192.168.117.4:5060: INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.117.4:5060;branch=z9hG4bK7bd677b8 From: WIRELESS CALLER sip:[EMAIL PROTECTED];tag=as37dd610c To: sip:[EMAIL PROTECTED];tag=as3af9dc41 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 INFO User-Agent: ISDN-NET Voip Gateway Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=500 --- I know it is not the receiving box messing things up as I get the same short short DTMF sound on a cisco IAD. Something is wrong with this packet but I just can't see it!!! Is there any rtp that gets sent, anyone know what the Content-length does? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO Duration=250
Where can I change the Duration length of an INFO packet? Content-Type: application/dtmf-relay Content-Length: 24 Signal=5 Duration=250 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
hi gurminder, are you using it on isdn line or pots line?On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote: HiDigium cards are compatible with indian telephony..I am using it.But there is problem I am facing to configure caller ID. What cidsignalling is used in india?RegardsGurminderOn 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem?Thanks in advance,AnkitP.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
I m using it on POTS line and will start with ISDN soon :-). Cheers Gurminder On 8/9/05, Ankit [EMAIL PROTECTED] wrote: hi gurminder, are you using it on isdn line or pots line? On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote: Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
where did u purchase ur card frm, im not able to find ne distributor of digium cards in india, and if i order it frm their site it will have to pay arnd 2k rs for shipping :( -ankit On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote: I m using it onPOTS line and will start with ISDN soon :-).CheersGurminderOn 8/9/05, Ankit [EMAIL PROTECTED] wrote:hi gurminder,are you using it on isdn line or pots line? On 8/9/05, Gurminder Arora [EMAIL PROTECTED] wrote: Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem?Thanks in advance, AnkitP.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
In india no distributer for digium cards If any body is going to us u can ask them to bring it. I got in that way -sandeep Ankit wrote: where did u purchase ur card frm, im not able to find ne distributor of digium cards in india, and if i order it frm their site it will have to pay arnd 2k rs for shipping :( -ankit On 8/9/05, *Gurminder Arora* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I m using it on POTS line and will start with ISDN soon :-). Cheers Gurminder On 8/9/05, Ankit [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi gurminder, are you using it on isdn line or pots line? On 8/9/05, Gurminder Arora [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Digium cards are compatible with indian telephony.. I am using it. But there is problem I am facing to configure caller ID. What cidsignalling is used in india? Regards Gurminder On 8/8/05, Ankit [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] info regarding hardware
Hi everybody, I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance, Ankit P.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] info regarding hardware
hi Digium card is compatible with the indian telephone line.I am currently using TDM400P(TDM11B) and this is working fine.It has one FXS and one FXO interface . rajeshAnkit [EMAIL PROTECTED] wrote: Hi everybody,I need a little clarification regarding the hardware to be used with asterisk. I want to setup an asterisk box to make calls through both internet and pstn, but i heard frm my friend (he was not sure) that digium cards are incompatible with indian telephony systems, is it so? If yes, then is there a way around this problem? Thanks in advance,AnkitP.S- It would be greatly appreciated if someone could provide a technical explanation to why digium cards are incompatible with indian (or anyother telephone system), i thought telephone network is same everywhere.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info on ACD in Asterisk
Title: Info on ACD in Asterisk Hello Sir, I have few clarifications as we are planning to work with asterisk. If you dont mind, please clarify the following:- Q1. Do Asterisk support ACD functionality? If Yes, can you give information on how to configure or work with ACD (and its usage). Q2. From the list of features listed in www.asterisk.org , I see Predictive dialler is listed under Telephony Services but Not ACD functionality under Call Features Q3. If ACD is not supported, how come Predictive Dialer is going to work? Please do clarify me. Thanks Best Regards, Ajay Kanth Ph: 9848880309 Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or [EMAIL PROTECTED] immediately and destroy all copies of this message and any attachments. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info abaut zaphfc
I'm trying to correct the cid for italy because when arrive a call cid display the number without the initial 0 and when i want to redial the missed call i can't because the number is wrong Thank's Tiziano
Re: [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Andres, the normative reference for abrorbing retranmissions is in RFC3261 -- INFO/RFC refers to it by telling all transaction handling is like for BYE requests. This is the specific piece of text from RFC3261 which explains why not absorbing retransmissions breaks the spec. Cheeers, -jiri 17.2.2 Non-INVITE Server Transaction ... Once in the Trying state, any further request retransmissions are discarded. ... On Mon, 23 Feb 2004, Andres wrote: Hi Jiri, I certainly welcome and applaud your comments and suggestions. But I could not continue to push this issue as an asterisk bug since the RFC did not back me up. Absorbing these SIP INFO retransmissions is more like a common sense thing/feature that should be implemented in asterisk rather than an RFC violation, since the RFC is quite vague. If anybody has the knowledge to implement this feature I can certainly help test it. Regards, Andres. Jiri Kuthan wrote: Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Gee, maybe I'm missing something, but the spec does not say that. The RFC actually says that when you send a final response, you are required to store that final response for 64*T1 seconds and retransmit the final response each time you receive the retransmitted request. (T1 = 500ms) Otherwise, you would be screwed if you you one and only final response was lost in transit. :) Jiri Kuthan wrote: Andres, the normative reference for abrorbing retranmissions is in RFC3261 -- INFO/RFC refers to it by telling all transaction handling is like for BYE requests. This is the specific piece of text from RFC3261 which explains why not absorbing retransmissions breaks the spec. Cheeers, -jiri 17.2.2 Non-INVITE Server Transaction ... Once in the Trying state, any further request retransmissions are discarded. ... On Mon, 23 Feb 2004, Andres wrote: Hi Jiri, I certainly welcome and applaud your comments and suggestions. But I could not continue to push this issue as an asterisk bug since the RFC did not back me up. Absorbing these SIP INFO retransmissions is more like a common sense thing/feature that should be implemented in asterisk rather than an RFC violation, since the RFC is quite vague. If anybody has the knowledge to implement this feature I can certainly help test it. Regards, Andres. Jiri Kuthan wrote: Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
On Tue, 24 Feb 2004, Clif Jones wrote: Gee, maybe I'm missing something, but the spec does not say that. I copiedpasted the wording bellow from RFC3261. The RFC actually says that when you send a final response, you are required to store that final response for 64*T1 seconds and retransmit the final response each time you receive the retransmitted request. (T1 = 500ms) Sure. That's not mutualy exclusive. Absorbing a retransmissing means you don't pass it to application, that's the point. If you already sent a reply, it is perfectly valid to resend it. -jiri Otherwise, you would be screwed if you you one and only final response was lost in transit. :) Jiri Kuthan wrote: Andres, the normative reference for abrorbing retranmissions is in RFC3261 -- INFO/RFC refers to it by telling all transaction handling is like for BYE requests. This is the specific piece of text from RFC3261 which explains why not absorbing retransmissions breaks the spec. Cheeers, -jiri 17.2.2 Non-INVITE Server Transaction ... Once in the Trying state, any further request retransmissions are discarded. ... On Mon, 23 Feb 2004, Andres wrote: Hi Jiri, I certainly welcome and applaud your comments and suggestions. But I could not continue to push this issue as an asterisk bug since the RFC did not back me up. Absorbing these SIP INFO retransmissions is more like a common sense thing/feature that should be implemented in asterisk rather than an RFC violation, since the RFC is quite vague. If anybody has the knowledge to implement this feature I can certainly help test it. Regards, Andres. Jiri Kuthan wrote: Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Andres Network Admin http://www.telesip.net
Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Serusers] Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Hi Jiri, I certainly welcome and applaud your comments and suggestions. But I could not continue to push this issue as an asterisk bug since the RFC did not back me up. Absorbing these SIP INFO retransmissions is more like a common sense thing/feature that should be implemented in asterisk rather than an RFC violation, since the RFC is quite vague. If anybody has the knowledge to implement this feature I can certainly help test it. Regards, Andres. Jiri Kuthan wrote: Andres, thanks for your reply. I beg to disagree, here are the arguments: 1) Having INFO is imho a useful thing: it allows elements out of the media path to control DTMF-based service logic. Otherwise, you will end up processing media which affects bandwidth and latency noticably and does not scale. 2) Apart from the out-of-order argument, reprocessing retransmissions is a bug worth fixing. It is responsibility of transaction layer to absorb UDP retransmissions and never let app see them. (Similarly like TCP does not pass retranmissions to apps.) I think there are more cases for proper transaction processing other than just DTMF/INFO. 3) out-of-order delivery may or may not be an issue: gnerally, one would need to mainain a kind of playout buffer like for RTP. O-o-o delivery does not matter to me personaly since I send DTMF/INFO in stop-and-go mode. (BTW, I think the text in the RFC is not entirely correct, re-INIVITE should not cause CSeq gaps. Nevertheless, the RFC does not prevent anybody from implementing an INFO playout buffer). -jiri On Sun, 22 Feb 2004, Andres wrote: Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
Hi Jiri, Been there. We switched from INFO to RFC2833 for this same reason. Take a look at: http://bugs.digium.com/bug_view_page.php?bug_id=0001033 Not only retransmissions are affected but out of order packets too. This behaviour can be partly blamed on the RFC: In addition, the INFO method does not define additional mechanisms for ensuring in-order delivery. While the CSeq header will be incremented upon the transmission of new INFO messages, this should not be used to determine the sequence of INFO information. This is due to the fact that there could be gaps in the INFO message CSeq count caused by a user agent sending re-INVITES or other SIP messages. Regards, Andres Jiri Kuthan wrote: I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Network Admin http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO/DTMF retransmissions in * not absorbed?
I'm wondering whether people know if there could be a problem with * receiving retransmissions of INFO/DTMF requests. I'm trying to play DTMF via INFO to *. If it takes a 200 reply too long to come back, the request is retransmitted. Whenever this happens, the IVR down in PSTN reports that the number sequence is incorrect. This makes me guess that * turns INFO retransmissions into new DTMF digits on the PSTN part. Does anybody have the same experience? Is it a known problem? Are there any patches? Thanks, -jiri -- Jiri Kuthanhttp://iptel.org/~jiri/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Fri, Oct 31, 2003 at 10:36:23AM +1100, Anthony Wood wrote: S-bus might be ISDN BRI ports, in which case Asterisk can plug in with an AVM Fritz (~110 euro) and chan_capi. OK, and would I be right in thinking that each Fritz!Card will support a maximum of 2 channels? Mind, I'd expect that to be plenty for the purposes of testing the system... http://www.telappliant.net/site2/mypbx_solution.htm And whilst I like the idea of a pre-configured appliance, I don't know if you get root access, etc. since we will need to write our own applications, etc. AGI (asterisk gateway interface??) is an application interface for Asterisk, which can use perl, C, php and probably other languages... Yep I'm aware of AGI, but assumed the scripts would need to run on the * box itself, hence my concern over root access. Either way, Telappliant haven't bothered to reply to my e-mails in 2 days... Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Fri, Oct 31, 2003 at 10:32:13AM +1100, Anthony Wood wrote: And the reverse is possible too, if you buy 2 E100 cards, you can plug your old PABX into the Asterisk server and set up with very minimal config so each proprietry handset can be used with Asterisk. Another interesting idea, but not really appropriate - I don't want to replace the current one - it'll be staying the way it is in this building - I'm only proposing to develop a PBX for the new building :) If you are going this route, you should consider a TE410P which will give you future options of T1 channel banks, extra E1 lines etc. Channel banks, from what I've gathered, are only useful if you want to use POTS analogue phones - is that correct? Whilst the call-centre staff are pleasant, they're not the brightest bunch, bless their fluffy little heads - so a digital phone with LCD, programmable buttons, and lights to show what lines have calls waiting to be answered, etc. is quite mandatory. This would seem to narrow my options to 'IP Phones' like the Budgetone / Snom ... I also see a lot about 'ADSI' support in * with regard to low-speed data signalling (1200 baud I believe) to analogue extensions - will a channel bank help here? What kind of ADSI phones are available at what sort of cost, since I've not been able to find any information :( Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info on UK ISDN30e?
Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Gavin Hamill wrote: Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? Cheers, Gavin. AFAIK ISDN30 is the right thing.. BT will provide an RJ45 port for you to connect the Digium card to.. As for the ISDN2's I don't know if its a good idea, especially since the 2 and 4 port BRI cards are very expensive.. Can't help you on the handset question.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On 30/10/03 14:38, Gavin Hamill wrote: Has anyone used ISDN30e in the UK with the Digium E1 cards? Many people. What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. It's more a question of how to implement the backup lines - they're fine for outbound calls, but it's backup for inbound lines that you really want. This is difficult to achieve, but you might be able to get BT to give you a hunt group that hits a pair of ISDN2s after looking through the E1 bank and failing to connect. It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. Do BT terminate the ISDN30e in a format that I can literally just plug into the Digium cards, or will I need some kind of adapter (whether electronics or even just a simple socket/plug changer)? You will be able to just plug it straight in (standard RJ45 termination). I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Can't answer this one, others? Many people/organizations have successfully deployed it, though. Be aware that it's currently not as easy to configure as many commercial PBXs, but it tends to be cheaper and more flexible. FAX support is also coming soon. :) Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? I doubt you can reuse proprietary handsets. Please provide more details (model/make). -- Alastair Maw MX Telecom www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Finally, are my options for handsets limited to IP phones via Ethernet, or analogue phones via a channel bank (and then to another Digium E1/T1 card), or is there the possibilty to re-use proprietary handsets from a previous PBX? One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Linus Magrathea Telecommunications (provider of IAX termination and origination services in the UK!) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
At 14:38 + 30/10/03, Gavin Hamill wrote: The problem begins in that I only have a very loose grasp of the telco world. Has anyone used ISDN30e in the UK with the Digium E1 cards? What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. You can of course get ISDN30 from other suppliers than BT. Some may try and present as DASS rather than Q931. You want Q931 otherwise you need to get a convertor box. Q931 is the RJ45 version that you just plug in to the line card. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. HTH f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:00:09PM +, Alastair Maw wrote: On 30/10/03 14:38, Gavin Hamill wrote: Has anyone used ISDN30e in the UK with the Digium E1 cards? Many people. That's reassuring to hear :) What options are there to stick on a couple of ISDN2's on top of that should we require some 'backup lines'. It's more a question of how to implement the backup lines - they're fine for outbound calls, but it's backup for inbound lines that you really want. Precisely - the outbound stuff we can route via CPS or any Internet SIP provider... We run a call-centre, so it's utterly crucial that the incoming number is always reachable via some method. This is difficult to achieve, but you might be able to get BT to give you a hunt group that hits a pair of ISDN2s after looking through the E1 bank and failing to connect. I think that'll probably be the route we take - I don't imagine management would be very keen on someone routing the call data to us via a network stream - that would require a lot of high-quality, high-expense bandwidth (i.e. UUNet in the UK) It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. I don't quite understand what you mean by this - we want to terminate the ISDN30e ourselves, and have a couple of ISDN2s also there 'just in case'. You will be able to just plug it straight in (standard RJ45 termination). Superb. I'm trying to gather some tangibility for the project - I see the first mailing list post in November 1999... when did the project start, and when was it first usable as a simple PBX? Can't answer this one, others? Many people/organizations have successfully deployed it, though. Ah now that continues in the theme of my question - What people? What organisations? What experiences / issues do they have to tell about the installation? There's the likelihood of a lot of great PR for Asterisk if the relevant parties would only put something in an e-mail :) I mean, I'd love to turn round and be able to say to the bigwigs: 'Hey look, Shell UK have converted their entire nationwide telephony system to Asterisk', but that isn't going to happen - but lots of positive reports from small businesses who have made the switch would (IMO) do Asterisk a power of good. Be aware that it's currently not as easy to configure as many commercial PBXs, but it tends to be cheaper and more flexible. FAX support is also coming soon. :) My previous experience with PBXs was a Siemens HiCom system, and even with the Windows-based config tool, I didn't find it terribly easy to configure. Indeed, I managed to kill the NVRAM on £1500-worth of VoIP card in the process - whoops! or is there the possibilty to re-use proprietary handsets from a previous PBX? I doubt you can reuse proprietary handsets. Please provide more details (model/make). We have an Inter-Tel Axxess unit with 32 extensions at present, and immediate plans for another 16. If I can hook the S0-bus Asterisk and some IP phones as a proof-of-concept, I'd be very happy. Cheers, Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote: I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. I think the point is that we've just about reached capacity on our 30 channels, and won't be in this building for much longer (basically as soon as we can get telephony + data into the new building) so rather than taking another ISDN30, just take a couple of ISDN2s to tide us over in the meantime... isn't eight the minimum no. of channel for a new ISDN30 installation? Q931 is the RJ45 version that you just plug in to the line card. OK thanks for that info, I'll keep it in mind.. I did see DASS2 refered to on BT's ServiceView price list, but didn't know what the alternative was called. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. Yes :) It was only a thought - I think they want to leave the current equipment in the current building in case we need to re-use it at a later date, or suddenly expand past capacity in the new building, etc. Many thanks for your time and comments! Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Very interesting :) There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told, which we run 4 faxmodems off - I'm not exactly sure /how/ they connect, tbh.. will need to check that out... Perhaps they're just 4 POTS analogue extensions... This would be the ideal testing ground for Asterisk (for me to learn on) since hopefully we could pass the incoming number to the S0-bus, hence Asterisk, hence any IP Phones we buy as a technology demo. The idea of taking a fresh ISDN30 and trying to get everything working from day 1 terrifies me :) We've looked at 'myPBX' from http://www.telappliant.net/site2/mypbx_solution.htm And whilst I like the idea of a pre-configured appliance, I don't know if you get root access, etc. since we will need to write our own applications, etc. As always, I'm open to ideas =) Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
Q931 is the RJ45 version that you just plug in to the line card. Q931 describes the protocol and not the line presentation. However, you do want to ensure that you ask for Q931 as although DASS/2 is an ISDN protocol, it isnt the same as Euro-ISDN and not supported by Asterisk. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
It's only worth doing if you're going to route them directly to some other kit, though, so Asterisk support for ISDN2 hardware is largely irrelevant here. I don't quite understand what you mean by this - we want to terminate the ISDN30e ourselves, and have a couple of ISDN2s also there 'just in case'. I think the person who replied meant that if you are having the lines as backup in case of failure, you should also be considering failure of the Asterisk equipment and therefore the backup lines should route to a different solution than the ISDN30e / Asterisk one. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
At 19:24 + 30/10/03, Gavin Hamill wrote: On Thu, Oct 30, 2003 at 03:24:24PM +, Fearghas McKay wrote: I would just get another ISDN30 and enable extra circuits as required, rather than add a couple lines here and there with ISDN2/BRI. I think the point is that we've just about reached capacity on our 30 channels, and won't be in this building for much longer (basically as soon as we can get telephony + data into the new building) so rather than taking another ISDN30, just take a couple of ISDN2s to tide us over in the meantime... isn't eight the minimum no. of channel for a new ISDN30 installation? you are asking for an extra bunch of channels, you already have the fibre probably, it may even have the capacity on it just not enabled. Wearing hat with scars on it - don't mess around doing it with ISDN2, the grief qoutient is too high, especially since you won't be there for long and can reuse the kit in the new location. And if you install ISDN2e you will have to take a 12 month contract and have large install costs - you might get away with Business Highway but your cost per channel is probably going to be higher than the ISDN install cost. If you are keeping the current system in place then just explain to account rep that you will be buying new lines from them in new site and keeping on most of the current lines, but what deal will they cut for you. Also remember you have a choice of ISDN providers, even if they are not in your area they can terminate on BT local loop. Q931 is the RJ45 version that you just plug in to the line card. OK thanks for that info, I'll keep it in mind.. I did see DASS2 refered to on BT's ServiceView price list, but didn't know what the alternative was called. You could probably reuse the handsets from the proprietory pbx, but it may be cheaper to save the time and complexity by justgetting new handsets, that would need an analysis. Yes :) It was only a thought - I think they want to leave the current equipment in the current building in case we need to re-use it at a later date, or suddenly expand past capacity in the new building, etc. Many thanks for your time and comments! nae problem. f ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 07:42:39PM -, Linus Surguy wrote: I think the person who replied meant that if you are having the lines as backup in case of failure, you should also be considering failure of the Asterisk equipment and therefore the backup lines should route to a different solution than the ISDN30e / Asterisk one. Ah of course :) I expect that would be an identical Asterisk installation, and manually switch over the RJ45 cable from box 1 to box 2 if box 1 fails.. Complete duplication of hardware - not very expensive in this case - the biggest expense would probably be the Digium E1 card, and that's a small price to enable such peace of mind! Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Info on UK ISDN30e?
On Thu, Oct 30, 2003 at 07:29:17PM +, Gavin Hamill wrote: On Thu, Oct 30, 2003 at 03:10:09PM -, Linus Surguy wrote: One option you might not have considered is connect your existing PBX to the back of Asterisk and thereby use it as a channel bank itself. Very interesting :) There *is* an 'S-bus' (which is the same as an 'S0-bus'?) I'm told, which we run 4 faxmodems off - I'm not exactly sure /how/ they connect, tbh.. will need to check that out... Perhaps they're just 4 POTS analogue extensions... S-bus might be ISDN BRI ports, in which case Asterisk can plug in with an AVM Fritz (~110 euro) and chan_capi. This would be the ideal testing ground for Asterisk (for me to learn on) since hopefully we could pass the incoming number to the S0-bus, hence Asterisk, hence any IP Phones we buy as a technology demo. The idea of taking a fresh ISDN30 and trying to get everything working from day 1 terrifies me :) We've looked at 'myPBX' from http://www.telappliant.net/site2/mypbx_solution.htm And whilst I like the idea of a pre-configured appliance, I don't know if you get root access, etc. since we will need to write our own applications, etc. AGI (asterisk gateway interface??) is an application interface for Asterisk, which can use perl, C, php and probably other languages... As always, I'm open to ideas =) A good philosophy. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO method and DTMF translation
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code:
line 3982
if (p-owner) {
if (strlen(buf)) {
if (sipdebug)
ast_verbose(DTMF received: '%c'\n, buf[0]);
event = atoi(buf); WHY?
if (event 10) {
resp = '0' + event;
} else if (event 11) {
resp = '*';
} else if (event 12) {
resp = '#';
} else if (event 16) {
resp = 'A' + (event - 12);
}
memset(f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
f.subclass = resp;
f.offset = 0;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(p-owner, f, 0);
}
On line 3986, any # or * digit I entered was translated to 0(zero). So
any apps depends on # for terminating (voicemail for example) won't
work.
My question is , why not take just buf[0]? why translate? my UA
always send something like d= (one digit) at a time.
--
Best regards,
Nguyen mailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] INFO method and DTMF translation
Hi.
The implementation is correct, I can use sip info
method to get all the DMTF, *,# included (eg voicemail
works great with sip info dtmf)
the line atoi(buf) is needed 'cause buf is a char, and
we need a int value to do the comparisons below that line.
and I don't see why they get set to 0 ... probably
'cause the INFO dtmf on your gw is broken.
sip info method describes dmtf as numbers (not chars) ,
so 0-9 are the digits, 10,11 are respectively *,#
and 12-16 are A,B,C,D
if you get translated to 0, I can assume that your gw
sends out # or * as char and not as numbers, as sip info
method requires.
matteo.
Il dom, 2003-10-12 alle 09:38, Nguyen Hoang Lan ha scritto:
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code:
line 3982
if (p-owner) {
if (strlen(buf)) {
if (sipdebug)
ast_verbose(DTMF received: '%c'\n, buf[0]);
event = atoi(buf); WHY?
if (event 10) {
resp = '0' + event;
} else if (event 11) {
resp = '*';
} else if (event 12) {
resp = '#';
} else if (event 16) {
resp = 'A' + (event - 12);
}
memset(f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
f.subclass = resp;
f.offset = 0;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(p-owner, f, 0);
}
On line 3986, any # or * digit I entered was translated to 0(zero). So
any apps depends on # for terminating (voicemail for example) won't
work.
My question is , why not take just buf[0]? why translate? my UA
always send something like d= (one digit) at a time.
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
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Re: [Asterisk-Users] INFO method and DTMF translation
Just for integration, look here
http://lists.digium.com/pipermail/asterisk-users/2003-July/016464.html
basically sip info dtmf are:
Event encoding (decimal)
_
0--90--9
* 10
# 11
A--D 12--15
Flash 16
matteo
Il dom, 2003-10-12 alle 09:38, Nguyen Hoang Lan ha scritto:
Hello guys,
I have searched high and low, but not found any information about
rules of using DTMF in SIP INFO method. Cisco has described something with
Signal=, but it look like this feature is dependent on implementors?
The problem is chan_sip.c cannot correctly translate received DTMF
digits, especially #,*. At least with my Antek EGW-804 gateway.
Looking into chan_sip.c, I found this code:
line 3982
if (p-owner) {
if (strlen(buf)) {
if (sipdebug)
ast_verbose(DTMF received: '%c'\n, buf[0]);
event = atoi(buf); WHY?
if (event 10) {
resp = '0' + event;
} else if (event 11) {
resp = '*';
} else if (event 12) {
resp = '#';
} else if (event 16) {
resp = 'A' + (event - 12);
}
memset(f, 0, sizeof(f));
f.frametype = AST_FRAME_DTMF;
f.subclass = resp;
f.offset = 0;
f.data = NULL;
f.datalen = 0;
ast_queue_frame(p-owner, f, 0);
}
On line 3986, any # or * digit I entered was translated to 0(zero). So
any apps depends on # for terminating (voicemail for example) won't
work.
My question is , why not take just buf[0]? why translate? my UA
always send something like d= (one digit) at a time.
--
Brancaleoni Matteo [EMAIL PROTECTED]
Espia - Emmegi Srl
___
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Re[2]: [Asterisk-Users] INFO method and DTMF translation
Hello Brancaleoni, Sunday, October 12, 2003, 4:39:32 PM, you wrote: BM Hi. BM The implementation is correct, I can use sip info BM method to get all the DMTF, *,# included (eg voicemail BM works great with sip info dtmf) BM the line atoi(buf) is needed 'cause buf is a char, and BM we need a int value to do the comparisons below that line. BM and I don't see why they get set to 0 ... probably BM 'cause the INFO dtmf on your gw is broken. BM sip info method describes dmtf as numbers (not chars) , BM so 0-9 are the digits, 10,11 are respectively *,# BM and 12-16 are A,B,C,D BM if you get translated to 0, I can assume that your gw BM sends out # or * as char and not as numbers, as sip info BM method requires. BM matteo. Thanks matteo. You are correct. My gw send out # and * instead of 10 or 11. So I have created a 'special' handling for this broken?! implementation. -- Best regards, Nguyenmailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] INFO: How the T410P sets the number of channels per span
After speaking with Martin @ Digium, I have the following answers. The driver Wct4xxp determines the number of channels by the signaling type set in the /etc/zaptel.conf file. For example if all the spans used b8zs,esf your spans would look like this: Span 1 Zap/1 - Zap/24 Span 2 Zap/25 - Zap/48 Span 3 Zap/49 - Zap/72 Span 4 Zap/73 - Zap/96 However if you have an E1 intermixed say on span 2 you channels would be: Span 1 Zap/1 - Zap/24 Span 2 Zap/25 - Zap/54 This is because the signaling s\is set to E1 Span 3 Zap/55 - Zap/78 Span 4 Zap/79 - Zap/102 So it is the driver that sets up the devices in the /dev/Zap directory, This could be a gotcha for newbies as the channels are all contiguous and the span separations are on numbers other than the standard 24 or 30. Of Course this is just an example and in a real life situation you would probable start with an E1 or end with an E1 instead of putting it smack dab in the middle. I hope the this helps. Message: 2 Date: Thu, 24 Jul 2003 01:07:54 -0400 From: Alex Lopez [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] T410P and zaptel.conf Reply-To: [EMAIL PROTECTED] One a t400p I know that I have 24 channels per port for a total of 96. = However the T410 card allows for E1 as well as T1 lines. How does it = determine how many channels per port. For a more specific question. Would the first Zap device on the second = port be Zap/25 or Zap/30 when using a T1?? I looked for docs on this but found nothing.. =20 Other questions: Is the electrical interface the same for a E1 as a T1?? How does the card know which is which? Is it by the span def in the = /etc/zaptel.conf file? Has anyone seen a technical document on this card??? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Info sip/h.323 interoperability
Hi all, I'm a student (my thesis work consist in testing interopearbility SIP/H.323) and I begin to work with asterisk in this days. I have to testing to SIP/H.323, since today I have used Vocal system, but there are some problem for this features. In the asterisk mailing list, in the next message I've seen an e-mail [Asterisk-Users] Cisco 7960 with Asterisk H.323 Shaun Ewing [EMAIL PROTECTED] Mon, 26 May 2003 21:56:42 +1000 I've red in that mail youhave over a hundred 7960s using Asterisk and chan_h323 , so my question is: Asterisk supports this interoperability ? I have done some test with Vocal to make calls from IP cisco Phone via sip/h323 translator to connect with Netmeeting or other h.323 end points... If this interoparability is supported which module I needs ? Thanks for attention, Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
