[asterisk-users] app_txfax / app_rxfax
Anyone got any clues or patches on how to make these work with the latest svn trunk. The only versions of app_txfax.c don't compile Julian. ___ --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] IP phone with 2 ethernet jacks
We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. Guido Hecken wrote: We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --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] H323
Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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] Re: SV: E61
On 26 Aug 2006, at 07:57, Martin Joseph wrote: Now, the fact you can't easily get these phone in the US, that's a conspiracy ;~) ... and if you take them to the US you realize you should have gotten a quad-band phone because your E60 can't deal with the common US frequency of 850 MHz, which European tri-bands don't have. My reception is bad pretty much wherever I am, using Cingular :( jens ___ --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] H323
Hi, i used to work ooh323 with my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )but ooh323 works fine with me thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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] Uptime Record?
Our MOH died, so I finally had to kill my * process and restart it. Interestingly, stop now didn't work. I had to kill the process. It used to work, but it had been up so long that it must have gotten corrupted somehow. Here is the show uptime before I killed it: Asterisk-A*CLI show uptime System uptime: 1 year, 24 weeks, 3 days, 10 hours, 1 minute, 33 seconds Last reload: 11 hours, 30 minutes, 49 seconds Asterisk-A*CLI Who says * isn't stable enough for prime time? At least it is on 1.0.3. ___ --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] GSM gateway and FXO ATA
On Sat, Aug 26, 2006 at 01:35:36AM +0800, Sam Tam wrote: WE can provide you with budget GSM Gateway if you are interested? which is commercial nope? wrong list again? could have been private Email? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] can not get ${LEN(VAR)} and greater than to work for me
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial) in new
stack
indicating that the number was not greater than 7.
if i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}=11]?dialout:nodial);
and dial the same 1 800 xxx
i get:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 1?dialout:nodial) in new
stack
indicating that the length of number dialed was equal to 11 digits.
so equal to works and greater than does not?
Can any one see what I am doing wrong?
* version 1.2.9.1
TIA
John M
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Re: [asterisk-users] Re: Attempt to setup paging and intercom
Thanks for your reply Steven.
I appears to me that that the extens in Intercom Group are patterns
requiring an initial underscore
but the extens in 2) One to Many Paging and 3) One to Many Intercom
are named extensions and should not have an initial underscore as
(mistakenly) shown.
That is,
exten = _**2
refers to an extension that was meant to be dialed as
star star 2 and should not have been preceeded with an underscore.
Is this also your understanding?
That said, the funtions do not work either way.
I put everything below (with underscores in 2) and 3) removed)
into extensions.conf and, under CLI, issued reload.
Is that the correct place?
Larry
Steven wrote:
I do not know if this breaks anything or not the way you have it, but you
should not have the underscore before the extension.
The underscore means that the following is an expression, where X=any single
digit and .=any number of digits.
I do not know if the underscore also interprets the * as something, or maybe it just gets stuck trying to figure out an expression
with no X nor .
Or this may not be an issue at all.
Larry Alkoff [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
This is my first attempt to setup intercom and paging for some
Grandview sip phones per instructions from Grandview.
I put the lines below in extensions.conf and did the CLI reload
command.
When I issue
**1 or **2 from a phone I get a 404 error.
Shouldn't that be ringing the 3 phones on my list?
The instructions are a little vague (to a newbie like me) and may
well be wrong.
Here is what I put in extensions.conf:
-- Stop reading here if not interested
; from: FAQ_Asterisk_Paging_for_GXP-2000.pdf
; Paging and Intercom:
;
; Grandstream Phone Configuration:
; Allow Auto Answer by Call-Info: Yes
; Turn off speaker on remote disconnect: Yes
; Note: Above configuration will allow GXP-2000 to auto answer a call
; when the call contains:
; SIP header Call-Info: answer-after=0
; And when the call hung up by the remote party,
; the phone will automatically on hook without alerting user with
; disconnect busy tones.
; Asterisk Configuration:
; ===
; Then you can set up Asterisk with following functions:
; 1) One to One Intercom
; ==
; You will first define a Macro and then use it in the one to one
intercom context:
[macro-pageext]
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for
ANY call
exten = s,2,SIPAddHeader(Call-Info: answer-after=0)
exten = s,3,Dial(${ARG1})
exten = s,4,NoOp() ; Add others here
exten = s,5, Hangup
exten = s,102,Hangup
[INTERCOM_GROUP]
exten = _*1XX,1,Macro(pageext,SIP/${EXTEN:1}) ;Page each extension
exten = _*1XX,2,Hangup
; Note: Above configuration will allow user intercom with any extension
; (using 1XX) by dialing *1XX.
; 2) One to Many Paging
; =
[One_Way_Page_GROUP]
exten = _**1,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**1,2,Page(${One_Way_Paging_List}|)
exten = _**1,3, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
; to all
; the extensions defined in variable One_Way_Paging_list
; which can be define as following:
One_Way_Paging_List = SIP/120SIP/122/SIP/100
; 3) One to Many Intercom
; ===
[Two_Way_Intercom_GROUP]
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(${Two_Way_Intercom_List}|d)
exten = _**2,3, Hangup
; Note: Above configuration will allow user to do two way intercom
to all the
; extensions defined in variable Two_Way_Intercom_List which can be
; define as following:
Two_Way_Intercom_List = SIP/120SIP/122/SIP/100
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Re: [asterisk-users] can not get ${LEN(VAR)} and greater than to work for me
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
John Millican wrote:
Hello all,
I am trying to test if the length of a dialed number is greater than 7. When
i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial) in new
stack
indicating that the number was not greater than 7.
if i use:
exten = 1,n,GoToIf($[${LEN(${numdial})}=11]?dialout:nodial);
and dial the same 1 800 xxx
i get:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 1?dialout:nodial) in new
stack
indicating that the length of number dialed was equal to 11 digits.
so equal to works and greater than does not?
Can any one see what I am doing wrong?
* version 1.2.9.1
Maybe string comparison because of the speech marks?
- --
Cheers,
Matt Riddell
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[asterisk-users] Asterisk Performance without RTP?
If Asterisk was used to set up and tear down calls, and using canreinvite allowing the RTP to pass from end-point to end-point, how many calls could Asterisk handle at once? I ask because I have been utilizing OpenSER but find myself constantly needing Asterisk to do this or that, and would like to move OpenSER into more of a Registration server, and letting Asterisk handle all of my calls I understand that the set up and tear down may be a tad slower, but programming (using AGI, etc.) would definitely outweigh the timing IMO. Thanks in advance. Kw ___ --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] Problem with Tycho Voicemail
Hi list! Im using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Somebody use that software any time ? have you the same problem ? Thanks ___ --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] Nobody is responding. Why? (Implement music on transfer)
Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are:1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.___ --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] Uptime Record?
On 8/26/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Who says * isn't stable enough for prime time? At least it is on 1.0.3. What kind of abuse does that box take? ___ --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] Nobody is responding. Why? (Implement music on transfer)
Use the m option at the end of the dial string. Google told me so.On 8/26/06, Crazy Boy [EMAIL PROTECTED] wrote: Hi friends,I did music on hold. How can we implement music on call transfer? I am unable to find any tutorial about setting up music on call transfer, when call is transferring from one extension to another extension. So, I have two doubts. They are: 1) How can I put music on call transfer (Not in music on hold)?2) Music on hold and Music on transfer are the same?Looking forward to your response. Thank you.Regards,Chandra. All-new Yahoo! Mail - Fire up a more powerful email and get things done faster. ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] getting SIP to listen on multiple ports
Is it possible to get sip to listen on two ports (say 5060 and 5061)? Maybe its not necessary, but I'm trying to get a PAP2 to work with 2 lines configured behind a Linksys router with NAT. I've noticed the default config in the PAP2 is to use 5060 for line 1 and 5061 for line 2. I'm guessing this is to assist in the handling of SIP through a NAT. If I try using 5060 for both lines I never see a registration for line 2. Any ideas? TIA ___ --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] H323
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeetingOn 8/26/06, atik khan [EMAIL PROTECTED] wrote:Hi,i used to work ooh323 with my asterisk. it gives better performance than otheroh323 or H323 comes with asterisk...i got H323 channel and oh323 with a lot of error.( like codecselection )but ooh323 works fine with methanksatikOn 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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 listTo 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] Re: getting SIP to listen on multiple ports
Please disregard this message. Evidently changing the port required a power cycle on the PAP2. On 8/26/06, Mr. Jones [EMAIL PROTECTED] wrote: Is it possible to get sip to listen on two ports (say 5060 and 5061)? Maybe its not necessary, but I'm trying to get a PAP2 to work with 2 lines configured behind a Linksys router with NAT. I've noticed the default config in the PAP2 is to use 5060 for line 1 and 5061 for line 2. I'm guessing this is to assist in the handling of SIP through a NAT. If I try using 5060 for both lines I never see a registration for line 2. Any ideas? TIA ___ --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] Uptime Record?
There aren't a lot of phones. There are 50-60 SIP phones and SIP connections to two Cisco PRI gateways. About 10,000 calls / month and about 15,000 mins of LD/month. I know when I started with *, I head how it had to be restarted every week and ours just ran and ran. Justin Tunney wrote: On 8/26/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Who says * isn't stable enough for prime time? At least it is on 1.0.3. What kind of abuse does that box take? ___ --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] can not get ${LEN(VAR)} and greater than to work for me
At 07:24 AM 8/26/2006, you wrote:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
and I dial an 11 digit number i.e. 1 800 xxx
i get this in the console:
Executing GotoIf(SIP/xxx-xxx-xxx-xxx-006ca720, 0?dialout:nodial)
in new stack
I'm guessing that you need to remove the quotes like this:
exten = 1,n,GoToIf($[${LEN(${numdial})}7]?dialout:nodial);
You know neither side of the expressions will never be empty so you
don't need them. You might also try something like:
exten = 1,n,GoToIf(${len({numdial:7})}?dialout:nodial)
or even:
exten = 1,n,GoToIf($[${numdial:7}=]?nodial:dialout);
Ira
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RE: [asterisk-users] Help compiling asterisk-addons on Debian?
Here is a detailed install guide for FreePBX but helps even if your not using FreePBX. http://powerontech.com/freepbx-on-debian.htm From: Christopher Aloi [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:09 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Help compiling asterisk-addons on Debian? Thanks for the tip!libmysqlclient12-devGot it done On 8/25/06, Rushowr [EMAIL PROTECTED] wrote: Do you have the development libraries installed too? I believe on Debian it's something like libmysqlclient From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Christopher AloiSent: Friday, August 25, 2006 8:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- --Christopher T Aloi-- ___ --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] IP phone with 2 ethernet jacks
I gotta put in a plug for my favorite phone the Aastra 9133i which also has BLF for each programmable button. Best all around reasonably priced business grade phone IMHO. -Original Message- From: Mario [mailto:[EMAIL PROTECTED] Sent: Saturday, August 26, 2006 2:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IP phone with 2 ethernet jacks We have used both IP501, IP601 (Polycom), Snom 320 and Snom 360. All of them are good phones with very good quality of voice and full of features. However, SNOM phones have a feature (missing from Polycom) that most of our customers really require: with SNOM phones you have leds for presence support that allow you to see which other extensions are busy (through the Asterisk Hint command). If this is important for you, you should really stay with Snom. Guido Hecken wrote: We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --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] Re: New Asterisk Voice Changer 0.4
To anyone having problems installing SoundTouch or libsoundtouch4c, I've improved the build system for libsoundtouch4c and updated the install instructions. Please let me know if you continue to have problems. http://www.lobstertech.com/code/voicechanger/ - Justin ___ --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] zap channel media volume
I've been struggling with this issue for over a year. I wish there were some kind of automatic gain control built in to set the rx/tx gain on the fly based on the volume of the two channels. Probably not realistic though. Is there other hardware other than digium's that better deals with this issue? Rich Adamson wrote: The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? ___ --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] Linksys PAP2 Ring Settings
That worked great!. I was using Ring_WaveForm and I guess it's case sensitive and the correct form should be Ring_Waveform. Thanks, Daniel On Aug 25, 2006, at 11:48 PM, Shanon Swafford wrote: This works for me on my SPA-3000 ver 3.1.10(GWd). Ring_WaveformTrapezoid/Ring_Waveform Then back to default. Ring_WaveformSinusoid/Ring_Waveform PAP2-NA shouldn't be any different. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Friday, August 25, 2006 6:27 PM To: Non-Commercial Discussion Asterisk Subject: [asterisk-users] Linksys PAP2 Ring Settings I have a few PAP2-NA that are being mass configured using the instructions on the wiki for the Sipura mass configuration. However, I need to make sure the following settings are in place as follow: Under the Regional Tab, I need the Ring Waveform to be Trapezoid instead of Sinuzoid and the Synchronized Ring to be Yes instead of No. I made an entry in the XML file for Synchronized_Ring which works just fine. However, no matter what I use for the Ring Waveform (Waveform, Ring_Waveform, Regional_Ring_Waveform), the setting is always the default (Sinuzoid). Does anyone know what the XML tag name/ settings need to be for changing the Ring Waveform? Thanks, Daniel ___ --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 ___ --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] Re: Re: Attempt to setup paging and intercom
I dug into my config (I do use paging over the phone but remember playing with
it) and came up with this reference.
-
exten = 5488,1,Page(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED])
[ext-pager]
include = ext-paging-custom
exten = _PAGE,1,SetVar(_ALERT_INFO=info=alert-autoanswer\;delay=1)
exten = _PAGE,n,Dial(SIP/${EXTEN:4})
exten = Debug,1,Noop(dialstr is LOCAL/PAGE${EXTEN:[EMAIL PROTECTED])
I do not remember if the LOCAL was required or not, but when I dial 5488, it
does page multiple phone.
Disregard the ALERT_IFO reference as it is specific to each phone type. We are
using Citel Handset Gateways to reuse old (yet
sturdy) NEC DTERM Series E phones.
We went with overhead paging because I did not want to even test how many
phones it takes to break paging over the phone. We have
250 phones witin the building.
There is also an issue of the fact that a current call is put on hold when you
page to it.
--
--
Steven
http://www.glimasoutheast.org
Larry Alkoff [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Thanks for your reply Steven.
I appears to me that that the extens in Intercom Group are patterns
requiring an initial underscore
but the extens in 2) One to Many Paging and 3) One to Many Intercom are
named extensions and should not have an initial
underscore as (mistakenly) shown.
That is,
exten = _**2
refers to an extension that was meant to be dialed as
star star 2 and should not have been preceeded with an underscore.
Is this also your understanding?
That said, the funtions do not work either way.
I put everything below (with underscores in 2) and 3) removed)
into extensions.conf and, under CLI, issued reload.
Is that the correct place?
Larry
Steven wrote:
I do not know if this breaks anything or not the way you have it, but you
should not have the underscore before the extension.
The underscore means that the following is an expression, where X=any single
digit and .=any number of digits.
I do not know if the underscore also interprets the * as something, or maybe
it just gets stuck trying to figure out an
expression with no X nor .
Or this may not be an issue at all.
Larry Alkoff [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
This is my first attempt to setup intercom and paging for some
Grandview sip phones per instructions from Grandview.
I put the lines below in extensions.conf and did the CLI reload
command.
When I issue
**1 or **2 from a phone I get a 404 error.
Shouldn't that be ringing the 3 phones on my list?
The instructions are a little vague (to a newbie like me) and may
well be wrong.
Here is what I put in extensions.conf:
-- Stop reading here if not interested
; from: FAQ_Asterisk_Paging_for_GXP-2000.pdf
; Paging and Intercom:
;
; Grandstream Phone Configuration:
; Allow Auto Answer by Call-Info: Yes
; Turn off speaker on remote disconnect: Yes
; Note: Above configuration will allow GXP-2000 to auto answer a call
; when the call contains:
; SIP header Call-Info: answer-after=0
; And when the call hung up by the remote party,
; the phone will automatically on hook without alerting user with
; disconnect busy tones.
; Asterisk Configuration:
; ===
; Then you can set up Asterisk with following functions:
; 1) One to One Intercom
; ==
; You will first define a Macro and then use it in the one to one
intercom context:
[macro-pageext]
exten = s,1,ChanIsAvail(${ARG1}|js) ; j is for dump and s is for
ANY call
exten = s,2,SIPAddHeader(Call-Info: answer-after=0)
exten = s,3,Dial(${ARG1})
exten = s,4,NoOp() ; Add others here
exten = s,5, Hangup
exten = s,102,Hangup
[INTERCOM_GROUP]
exten = _*1XX,1,Macro(pageext,SIP/${EXTEN:1}) ;Page each extension
exten = _*1XX,2,Hangup
; Note: Above configuration will allow user intercom with any extension
; (using 1XX) by dialing *1XX.
; 2) One to Many Paging
; =
[One_Way_Page_GROUP]
exten = _**1,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**1,2,Page(${One_Way_Paging_List}|)
exten = _**1,3, Hangup
; Note: Above configuration will allow user to one way page(broadcast)
; to all
; the extensions defined in variable One_Way_Paging_list
; which can be define as following:
One_Way_Paging_List = SIP/120SIP/122/SIP/100
; 3) One to Many Intercom
; ===
[Two_Way_Intercom_GROUP]
exten = _**2,1,SIPAddHeader(Call-Info: answer-after=0)
exten = _**2,2,Page(${Two_Way_Intercom_List}|d)
exten = _**2,3, Hangup
; Note: Above configuration will allow user to do two way intercom
to all the
; extensions defined in variable Two_Way_Intercom_List which can be
; define as following:
[asterisk-users] Re: IP phone with 2 ethernet jacks
Along the same lines as this question... Are there any Voip phones that have dual gigabit ethernet ports? ___ --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] Problem with Tycho Voicemail
Sergio R. D'Ippolito wrote: I’m using Tycho software to see my voicemail, y can see de detail from the message but i cant hear de message. Please send me or post here: - Client Version / os platform - Server Operating System - HTTP Server + php version - which version of the scripts you downloaded - your vmconfig.php config file without passwords! - if possible, look into your http server log for errors There is a known issue with sound not working on some Linux x86 platform not working reliably. ___ --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] Re: GSM gateway and FXO ATA
On 2006-08-25 10:35:36 -0700, Sam Tam [EMAIL PROTECTED] said: Hello WE can provide you with budget GSM Gateway if you are interested? Sam Hey Scumbag, How many timed do you need to be told that this isn't the place to sell your wares? Please Stop it! ___ --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] Re: GSM gateway and FXO ATA
On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over Grandstream HT488 ATA. snip Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. 3) The device hangs and reboots itself occasionally. This is all just an FYI. Marty PS I did test with the latest HT-488 firmware and all issues were still present. ___ --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] zap channel media volume
If one would visit with knowledgeable transmission engineers that work full time in the telephone industry, one would find telephony standards that govern exact transmission levels at each point throughout a country's telephone network (including the long distance facilities, pbx trunk loss, CO switch loss, etc). The only variable in those standards are the end user loops, which varies due to the length of the loop and other mostly uncontrollable and/or variable factors. The individual telephone companies oftentimes have internal transmission standards that govern what is or is not acceptable in terms of end user pstn loops. Practically all US telcos of any size force their installers to measure the transmission loss for every new installation, and oftentimes on any repair call. Asterisk's pc-based analog I/O cards totally ignores those standards. So, an automatic gain control would be nice but it would really be a work around for other root-cause / design problems. In testing various analog pstn I/O cards, I've found the sangoma A200D card (with hardware echo canceler) to be the best pstn analog interface on the market that address both the echo and transmission level issues for the longer higher-loss pstn loops. Transmission levels are still a little bit low but very usable. JD Austin wrote: I've been struggling with this issue for over a year. I wish there were some kind of automatic gain control built in to set the rx/tx gain on the fly based on the volume of the two channels. Probably not realistic though. Is there other hardware other than digium's that better deals with this issue? Rich Adamson wrote: The root cause of the low volume problem is the result of software echo cancellation software, and its need to insert a noticeable loss. If I recall correctly, the wctdm.c driver has a statically defined loss value of something like -6 db that is loaded into the TDM400 chipset at driver load time. Ordinarily, that loss is not all that noticeable. But, if your pstn line is rather lengthy (greater then about 5db worth of loss), the two loss values become very noticeable and marginal to users. There is no known fix or workaround. The low audio becomes even worse when a pstn caller leaves a voicemail and the user calls in via the pstn to retrieve his voicemail. The voicemail gain setting was intended to be sort of a workaround, but its marginal at best. JD Austin wrote: I've been fighting with this issue for over a year. There are several threads here talking about it: Digium Zaptel volume issues setting of volume Low volume/audio problems on TDM400 card increase the volume ? There is one thread (Voicemail volume adjustment) that give me hope that this can be fixed that mentions adding |usg(10) to the dial command to increase the gain. I'm still a novice at the inner workings of asterisk so I'm hoping one of the gurus on the list will figure this out eventually. JD Hi all, we do have the following configuration (non-Asterisk PBX) - T1 - ZAP (Asterisk PBX) - ZAP - T1 - (GSM Gateway) - GSM Enduser The call is originated on the (non-Asterisk PBX) - gets send over a T1 connection to the asterisk server (which does least cost routing) - the asterisk server then does send the call over a GSM Gateway into the world... The Problem we do have is - that the Users behind the non-Asterisk PBX are complaining about low volume media if the the calling through the gateway (if the are calling mobiles...). So i have started to raise the rxgain value for the connection between the asterisk box and the GSM Gateway, this does work quite well - but not really perfect. The ringback (not locally generated - does come from the GSM Provider) does get terrible loud - as soon as the callee is connected - the speech is nearly not hearable because it has such a low volume. The ringback is EARLY MEDIA - if i am right - and the speech is normal MEDIA. So, is it possible to set different gains for EARLY MEDIA and normal MEDIA ? Does anyone else have had this problem ? ___ --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] Re: GSM gateway and FXO ATA
Martin Joseph wrote: On 2006-08-22 01:59:09 -0700, Tomislav Parčina [EMAIL PROTECTED] said: Hi list! I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over Grandstream HT488 ATA. snip Personally I found the FXO port on the HT-488 to unworkable except as a backup for power outages. I found several problems with it. 1) serious echo issues (I have a long loop). 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. 3) The device hangs and reboots itself occasionally. This is all just an FYI. Marty PS I did test with the latest HT-488 firmware and all issues were still present. I'd agree with the above 1000%. It should be taken off the market. ___ --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] 7970 'LoadID incorrect' problem
Does ANYONE have any clues?
This is annoyng me no end :-(
Thanks
- Original Message -
From:
Paul A Brown
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, August 25, 2006 5:22
PM
Subject: [asterisk-users] 7970 'LoadID
incorrect' problem
Hi,
Just trying to setup my 7970 with latest SIP
image (SIP70.8-0-3S)
I referenced the page
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
And used the following as my
SEPmac.cnf.xml
devicedevicePoolcallManagerGroupmembersmember
priority="0"callManagerportsethernetPhonePort2000/ethernetPhonePort/portsprocessNodeName/processNodeName/callManager/member/members/callManagerGroup/devicePoolversionStamp{Jan
01 2005
00:00:00}/versionStamploadInformationSIP70.8-0-3S/loadInformationaddOnModules/addOnModulesuserLocalenameEnglish_United_States/namelangCodeen/langCode/userLocalenetworkLocale/networkLocaleidleTimeout0/idleTimeoutauthenticationURL/authenticationURLdirectoryURL/directoryURLidleURL/idleURLinformationURL/informationURLmessagesURL/messagesURLproxyServerURL/proxyServerURLservicesURL/servicesURL/device
But I get the 'LoadID incorrect'
error
How do I find the correct LoadID?
I simply reset the phone everytime with **#** in
settings
Thanks
Paul
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Re: [asterisk-users] 7970 'LoadID incorrect' problem
Does ANYONE have any clues? Only played with 7940 and 7960, but I will try to help since nobody comes forward loadInformationSIP70.8-0-3S/loadInformation Shouldn't that be something like P0S3-08-2-00 ? ___ --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] 7970 'LoadID incorrect' problem
- Original Message - From: Time Bandit [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 26, 2006 10:37 PM Subject: Re: [asterisk-users] 7970 'LoadID incorrect' problem Does ANYONE have any clues? Only played with 7940 and 7960, but I will try to help since nobody comes forward loadInformationSIP70.8-0-3S/loadInformation Shouldn't that be something like P0S3-08-2-00 ? ___ Thanks Going by all the examples posted (for earlier versions of the 7970 SIP image) you use the SIP name as there is no POS file for the 7970 :-( Thanks anyway ___ --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] ANNOUNCEMENT: Asterisk-Java 0.3-m1 released
Asterisk-Java 0.3-m1, a Java library for Asterisk PBX integration, has been released. The Asterisk-Java package consists of a set of Java classes that allow you to easily build Java applications that interact with an Asterisk PBX Server. Asterisk-Java supports both interfaces that Asterisk provides for this scenario: The FastAGI protocol and the Manager API. The 0.3-m1 milestone release focuses on ease of use and provides the new org.asteriskjava.live package that takes care of the lowlevel action and event handling of the Manager API and offers an intuitive API for Java developers. Asterisk-Java has been updated to take advantage of the new features of Java 5.0 and therfore requires a Java Virtual Machine of at least version 1.5.0. Asterisk-Java is used in several commercial environments and by the following Open Source projects: * Asterisk-JTAPI JTAPI implementation for Asterisk. http://asterisk-jtapi.sf.net/ * Asterisk-IM A plugin for the Jive Messenger XMPP (jabber) server. It provides integrated presence between your IM client and phone, notification of incoming calls by IM and originate calls from supported IM clients. http://www.jivesoftware.org/asterisk-im/ * Asterisk Desktop Manager (ADM) A desktop application that will allow for automatic on-call volume reduction, one click dial from clipboard, integrated phonebook and more. http://adm.hamnett.org/ Asterisk-Java is available under Apache 2.0 license at http://asteriskjava.org signature.asc Description: OpenPGP digital 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
RE: [asterisk-users] getting SIP to listen on multiple ports
I use a PAP2 device to register both lines to my Asterisk server ... Both the server and the PAP2 are inside my firewall and are on the same IP subnet ... In Asterisk's SIP.CONF I have the second line setup expecting SIP to use 5061 ... One rings a wireless phone while the other rings a CO line on my ancient KSU ... Works fine ... Bottom line is the PAP2 and the service registered to have to agree to the port being used ... Both lines are being accessed using the same IP address so you must designate unique ports to keep them separate ... I use this config every day as described above and know it works ... I suspect this would also work across NAT but the service you are registering to will have to expect you to be talking on 5061 instead of the default 5060 ... Not tough if you are controlling the config but might be a problem if you are trying to register to the same commercial SIP provider twice ... You will need to ask your provider to set up his side to expect you on 5061 if this is the case ... I have not tried this but can think of no reason why it would not work if the provider was willing to cooperate ... You can also use the PAP2 device to register to two separate providers ... But because the PAP2 uses a single IP address, the only way it can keep the two lines separate, is to use a unique port for each line ... So if I wanted to register to an external provider, I would set that channel up to use the default 5060, then use the second channel to register to my internal Asterisk system where I have the flexibility to configure the port expected to 5061 ... I have experimented with this config and know it works ... Hope this clears up the question for you ... If not let me know off list and I will see if I can take some screen shots of my PAP2 config and get together a sample of my SIP.CONF file for you to use as an exaple setup ... G.Hendershot -Original Message- From: Mr. Jones [mailto:[EMAIL PROTECTED] Sent: Saturday, August 26, 2006 12:33 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] getting SIP to listen on multiple ports Is it possible to get sip to listen on two ports (say 5060 and 5061)? Maybe its not necessary, but I'm trying to get a PAP2 to work with 2 lines configured behind a Linksys router with NAT. I've noticed the default config in the PAP2 is to use 5060 for line 1 and 5061 for line 2. I'm guessing this is to assist in the handling of SIP through a NAT. If I try using 5060 for both lines I never see a registration for line 2. Any ideas? TIA ___ --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] determining meetme user number
Hi,Is there a way to determine the MeetMeAdmin User number?I am using the MeetMeAdmin function from within the dialplan.I would like one of my admins to be able to drop out of the conference and be able to kick the last user that joined the conference. I believe that I can do this using:MeetMeAdmin(confno|k|userno)keeping track of the confno is easy since I created it,but I don't know how to determine the user number of the last person that joined the conference. Is there a way to store this in a variable before they join the conference? Or perhaps a way to detect the last user to join the conferences number?Cheers,Simon ___ --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] ticks in the pstn side audio
I am fighitng with this problem since last week. We use sipura 2100 ATA configured with rtp lenght about 20ms. Asterisk is connected to our upstrim using pri (Sangoma aft104d) During the call the pstn side hear a lot of ticks, I changed all kind of jitter buffer into the ata and patched asterisk to have jb on sip too..but the issue has not been fixed. I am pretty sure that the ata are fine becuase If I use a Mediatrix 2631 instead of asterisk, the audio quality is perfect. So I am little bit lost about it. Any suggestion? Giving the fact the problem is present only on the pstn side it has to do with the rtp stream exiting the ata or entring the asterisk box. Regards ___ --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] Re: IP phone with 2 ethernet jacks
On Sat, 2006-08-26 at 12:53 -0700, Martin Joseph wrote: Along the same lines as this question... Are there any Voip phones that have dual gigabit ethernet ports? Was wondering about that myself. I've wired every room with gb (for speeding up nfs) and i hate to loose that speed, because an ip-phone only has 100Mb switch internally. And i'm not fond on an extra cable or an external switch. ___ --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] Will Asterisk work with Exchange 2007 UM?
On Fri, 2006-08-25 at 14:50 -0400, [EMAIL PROTECTED] wrote: I'm faced with the need to create forensic test data for an Exchange 2007 server with unified messaging. Microsoft has a list of tested PBX and IP gateway products that are known to work (below) but I'd prefer to use Asterisk if possible. From everything I've read it appears that since Exchange uses SIP over IP and Asterisk uses SIP over UDP this will not work. I don't have a lot of experience with Asterisk but I was wondering if anyone knows of a plan to allow Asterisk to run SIP over IP or if there are any SIP gateways that will make this conversion. You can use (Open)SER or try the Asterisk (S)RPMs at http://www.laimbock.com/asterisk/ which have SIP TCP functionality. 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] Re: SV: E61
On 2006-08-24 06:32:27 -0700, Jon Schøpzinsky [EMAIL PROTECTED] said: I also have this phone, and have stumbled in to the same problem. I just think that it isn't in nokia's interest to change this, as it forces consumers to have some sort of local hardware, that (possibly) only the telecom provider can give them. This forces the users away from using cheaper services. Nokia makes a load from the telecom operators around the world, and are not interested in pissing them off, by letting their users bypass their price structure. Just my 5 cents. This is a bogus non-issue. Your system isn't configured right or the phone is set wrong. I have used my E60 from many locations on NATS outside the local LAN (which is also a NATTED config). I think the it's a conspiracy thing is a red herring. Now, the fact you can't easily get these phone in the US, that's a conspiracy ;~) Marty Marty, I was not going to get it based on what people said about the E61 and the NAT issues. Is this false ? I was thinking of getting it for when I travel to Israel. There seems to be a lot of open wifi connections all over the country there. Also how is the radio for the wifi on it ? ___ --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] Realtime and hints
You have me there. The onlu thing I can think of is to offer some one money to help build it for you. I had an issue with the p option in the dial command and paid some one on the developers list to patch it for me and then had him add it to SVN. Asterisk as a whole I think is good. It needs the improvements that we want it to have so maybe dare I say pay some one to modify the code. (Although I remember you saying a while back that you work for some one else and they may have insisted that you use asterisk because its free and some people seem to demand everything for open source and not pay a pennie). Dovid - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 12:54 PM Subject: RE: [asterisk-users] Realtime and hints We have bounced around various methods of efficient provisioning for months, and unfortunately none of them seem to be a very good solution. In regards to building static files, downloaded from the db: 1. Every time a customer makes a change, we have to download files again, and do a 'reload' again. If several customers are doing it at the same time, this could cause many downloads and reloads together. This doesn't seem like a good solution. We could batch them together, and do them at specified intervals, but then we have to tell customers via our web management interface that they have to wait X number of minutes for their findme/followme call routing changes to become effective. Not very nice. 2. How do you represent the files in the database? Do you store every line as a record? We tried building a tiered hierarchial structure of roles, clusters, hosts, files, contexts and elements for flexibility, but even with MySQL consultant help, it became very complicated. 3. I'm sure I've forgotten some stuff. In regards to using realtime: 1. You can't store BLF in realtime. 2. Realtime doesn't support ex-girlfriend logic. 3. You still need to use the 'include =' statement in the dialplan. This means your still going to have to make edits to the config files anyway from time to time, even with realtime. 4. The data as stored in the db is hard to manipulate for a Web Developer who doesn't know the inner workings of Asterisk. In my mind, provisioning and management are two of Asterisk's BIGGEST challenges. We've been stewing over it for a long time. Doug. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints Ok. Now I understand. What you can do is put it in a db and whenever you make changes you have asterisk grab the info from the db and put it into a file and reload asterisk. Of course asterisk supporting it would be a lot easier. Maybe you can get some one on the devel. list to do it. I also wondering if they will ever have support for complete real time of contexts. So I dont have to put them in the static files. I have a friend that when he has time will patch asterisk to look up a new table in the db with a list of contexts so that each time u create a new one you can just add it to the DB. Dovid - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 25, 2006 11:09 AM Subject: RE: [asterisk-users] Realtime and hints If you have to put one of the lines in extensions.conf, then you completely lose all the advantages that realtime gives you. You might as well just put both in extensions.conf now, as any change to the hint, or an addition, deletion etc is going to require a database change which is ok, but also an edit of the file and a subsequent asterisk reload. -Original Message- From: Dovid Bender [mailto:[EMAIL PROTECTED] Sent: Friday, August 25, 2006 7:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime and hints I dont know why it is working but it is. My first line I have in extensions.conf and the second I have in MySql. - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, August 24, 2006 1:00 PM Subject: RE: [asterisk-users] Realtime and hints I don't see how that helps. If you have a portion of the hint still in extensions.conf, then what use is the database? -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, August 24, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Realtime and hints That's what he was gettin at. Take the second line out, and put the first priority in the database. On
Re: [asterisk-users] Nobody is responding. Why? (Implement music on transfer)
You first step is going to be figuring out its an xfered call which
you can do by checking the ${BLINDTRANSFER} variable.
Then you just add the m option in the dial command.
http://www.voip-info.org/wiki/view/BLINDTRANSFER
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+dial
On 8/26/06, Crazy Boy [EMAIL PROTECTED] wrote:
Hi friends,
I did music on hold. How can we implement music on call transfer? I am
unable to find any tutorial about setting up music on call transfer, when
call is transferring from one extension to another extension. So, I have two
doubts. They are:
1) How can I put music on call transfer (Not in music on hold)?
2) Music on hold and Music on transfer are the same?
Looking forward to your response. Thank you.
Regards,
Chandra.
All-new Yahoo! Mail - Fire up a more powerful email and get things done
faster.
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[asterisk-users] Can the codec/format for name/greeting in voicemail be changed?
Sorry to badger everyone on the list but I never heard from even a single person on this so felt maybe I'll repeat it, just in case, it got unnoticed. Any ideas if it's possible to either record greetings/names in a different format than GSM OR be able to convert these voicemail subscriber greetings in my database to some other format? This is if I'm storing the voicemail and all greetings/etc in a SQL Server using Realtime. Thanks so much \R ___ --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] Call Max Time
Hi All,Could anyone give me idea, How i can set Call Max Time, so in pariticular time the call should disconnect automatically.I will be appriciate for your helps.Abdul Get your email and more, right on the new 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
RE: [asterisk-users] Call Max Time
Set(TIMEOUT(absolute)=seconds) Change seconds to the number of seconds you want to allow a call to last From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AbdulSent: Sunday, August 27, 2006 1:21 AMTo: Asterisk-Users@lists.digium.comSubject: [asterisk-users] Call Max Time Hi All,Could anyone give me idea, How i can set Call Max Time, so in pariticular time the call should disconnect automatically.I will be appriciate for your helps.Abdul Get your email and more, right on the new 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] hint for Hold
Hi, hint is used to monitor the status channels by using extensions in the dialplan. When an IP phone holds a call, there aren't any extensions sent to Asterisk. Does anyone know how I could monitor Hold ? For example, when an IP phone holds an existing call, the button on the phone that monitors the Hold blinks. regards, Kwang Mien ___ --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
