Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
Hi, 1. Is your WiFi network dedicated to VoIP or shared with data applications ? How was it designed ? For people using WiFi with a laptop, you propably don't need to have dense WiFi cells as moving from one cell should be scarce. With hand phones, those cells should overlap as it becomes very likely users would to move from one location to another while on the phone. 2. From this list, the WiFi hardphones which got only positive answers where Siemens Gigaset SL75 and Nokia EXX Series. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MSSQL ODBC Connections
hi, I want to create connection using odbc for mysql i have used cdr_odbc module for that. but when asterisk insert record to my mysql database arise "segfault error". any suggetion, pls give me tnks Bhrugu Mehta ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get IP address of an incoming or outgoing SIP call
I need it when the SIP calls comes to me in some context in a dialplan. The only way I can think of is right now which is kind of messy is to do 'sip show peers' parse the output in some agi script , get the ip from there. On Nov 26, 2007 1:53 AM, Johnny Tam <[EMAIL PROTECTED]> wrote: > > if you run asterisk in verbose mode.. asterisk -vvr then you will see > ips being made in and out in the log. > > > > Date: Mon, 26 Nov 2007 01:20:48 -0500 > > From: [EMAIL PROTECTED] > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] Get IP address of an incoming or outgoing SIP > call > > > > > > Hi * Users, > > > > What is the way from the dial-plan to get the IP address of an > > incoming or outgoing SIP call? I can see the IP address of the SIP > > call using 'sip show peers' from the CLI. > > > > Thanks > > > > Regards > > > > -- > > Arpit Mehta > > Graduate Student > > Department of Computer Science > > Columbia University > > > > Tel: 1-646-387-5998 > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided 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] Agi manager session.
Hi all, I am having requirment to display the status of the users continuously updated, for that I am having a php script which checks the status channels every seconds using the AMI .Now for this task the manager logs on and off every second. So, can any one help me to make manger just log in once and remain logged in Thanks in advance. ___ --Bandwidth and Colocation Provided 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] Get IP address of an incoming or outgoing SIP call
Hi * Users, What is the way from the dial-plan to get the IP address of an incoming or outgoing SIP call? I can see the IP address of the SIP call using 'sip show peers' from the CLI. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing DTMF upon call answering
On Sunday 25 November 2007 21:55:30 Robert McNaught wrote: > Does anyone know of a way to wait x number of seconds, then send the > DTMF digits as audio once the call is answered. Check out the M() parameter to Dial. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] passing DTMF upon call answering
Hi,
I am trying to write an application which sends DTMF tones once the
called party answers the call from asterisk.
>From the way I understand asterisk dialplans work - the below example
will NOT work as the dial application does not finish and move onto
the next priority once the call is answered. I tried this and
confirmed my thoughts.
exten => x,1,Dial(SIP/,20)
exten => x,n,SendDTMF(1234)
The asterisk book describes the D flag in the Dial application:
D([ called ][: calling ])
Sends DTMF digits after the call has been answered, but before the
call is bridged.
The called parameter is passed to the called party, and the
calling parameter is
passed to the calling party. Either parameter may be used individually.
Trying this in a dialplan:
Doing exten => _NXXNXX,1,Dial(SIP/CARRIERNAME/1${EXTEN},20,D(123))
results that you only hear 1 DTMF digit once the call is answered -
The asterisk CLI shows that 3 were send - I am assuming that the
difference is the time it takes for the audio to start streaming.
However, I was expecting to hear nothing, as the description in the
book above states that the DTMF is passed before the channels are
bridged (I understand this is around half a second after the call
signalling is answered)
Does anyone know of a way to wait x number of seconds, then send the
DTMF digits as audio once the call is answered.
Many thanks
Robert
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Re: [asterisk-users] MSSQL ODBC Connections
Thanks mate, this helped a lot On Nov 24, 2007 4:40 PM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Saturday 24 November 2007 14:48:02 Robert McNaught wrote: > > Does anyone know if it is possible to use the same database and single > > ODBC connection to do both CDR recording with cdr_odbc and dialplan > > routing based on func_odbc. > > In 1.4, no. The reason is, cdr_odbc was written prior to res_odbc and > therefore does not use its connections. So cdr_odbc attempts to make its > own connection to the database. > > Of course, if you're using MS SQL Server, that's what you want, because > concurrency is what disallows the use of a single connection. > > > I have both res_odbc.conf and cdr_odbc.conf pointing to the same DSN in > > odbc.ini > > > > I am starting to think that this limitation in having a single > > connection would stop this being possible in asterisk - does anyone know > > otherwise? > > There is no limitation in 1.4 of having a single connection, as long as you > set pooling=yes. So I don't understand your question. > > Please understand that odbc.ini doesn't set up a connection, only the > connection parameters. res_odbc can (and does) create multiple connections > based upon those parameters. > > -- > Tilghman > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Record() function] Script stops if user doesn't hit # after msg
On Sun, 25 Nov 2007 19:03:41 -0500, Doug Lytle <[EMAIL PROTECTED]> wrote: >What exactly are you trying to do? If a user hangs up during your >Record, it'll go directly the the h extension if it exists. Ah, didn't know about this extension :-/ I assumed Asterisk would go on to the next line in the script after executing Record(). I'll experiment and report back. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [1.4 - Record] How to tell if user did leave a msg?
On Fri, 23 Nov 2007 12:38:45 -0500, "Baji Panchumarti" <[EMAIL PROTECTED]> wrote: > Sometimes I use "press 1 to leave a msg" to reduce the > number of dead air msgs from callers. Good idea. BTW, making changes to zapata.conf did allow the Zaptel, and hence */Record to tell if the user hung up during the prompt. Just shows that regional settings can make a significant difference. It'd be cool if there were a site that gave newbie-friendly hints of telecom settings for such and such country. Thx. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Record() function] Script stops if user doesn't hit # after msg
Vincent wrote: > Hello > > I noticed something nasty with the Record() function: If the > user either hangs up during the prompt (ie. doesn't leave a message at > What exactly are you trying to do? If a user hangs up during your Record, it'll go directly the the h extension if it exists. I do this for recording pages, they record their page and then just hang up. It then gets played back over the PA system. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
For campus installations such as this, you may want to look at Polycom (Spectralink) phones. They are more expensive but are designed for tough environments and are of better quality than any of the consumer-oriented phones. Asterisk should be fine for an installation of this size, no need for SER. You can even set this up without registering the endpoints because they will be on a LAN and you can use static LAN IPs for each individual phone. Beyond technical considerations, you will have to be very aware of practical matters such as broken phones and stolen phones. In hospitals you can pretty much bet on replacing your total phone pool every year, so when installing a system like this you are not buying 100 phones, you are buying 100 phones PER YEAR. In a hotel I can imagine it would be even worse in terms of stolen phones. - Original Message - From: <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Sunday, November 25, 2007 3:26 PM Subject: [asterisk-users] Recommendations for 100 Wifi SIP phone setup > Hi all, > > > Im preparing a quote for a 5 Star hotel, planning to have around 100 > SIP Wifi phones for PBX operations running on 100 AccessPoints. > Network is running in ARUBA Networks - AP70 access points. > > The initial recommendation is to go for Hitachi Wifiphones, but i > would like to know from the group the recommendations. Im planning to > put up Asterisk as the PBX, Please advice me the do's and donts as i'm > not experienced on such heavy installation which are mission critical. > I had been using asterisk on small profiles and this would be my first > Pro setup with wifi handsets if all goes as planned. > > the Key Questions are > > Is Asterisk good enough? or do we need a another Proxy like SER? > > What is the experience with Hitachi Wifi phone's? Any specific Issues? > > Any such installations done? Please do a detail > > Looking for experiences.. > > Thanks > > Sunil Charly > Manager - Business Planning > KOLTELECOM > > ___ > --Bandwidth and Colocation Provided 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] [Record() function] Script stops if user doesn't hit # after msg
Hello I noticed something nasty with the Record() function: If the user either hangs up during the prompt (ie. doesn't leave a message at all), or does leave a message but forgets to hit the # key at the end... Asterisk stops right there, so the rest of the script doesn't run: [internal] exten => 777,1,Playback(leave_msg) exten => 777,n,Record(/tmp/test.wav,3,30) ;Verbose() not run exten => 777,n,Verbose(Here we are) Am I doing it wrong? Is there a way to solve this? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
My number one recommendation is be VERY VERY Careful. You could be selling the biggest nightmare to you and the customer ever. I have tried almost all the wifi sip phones and they are ALL sub par. Range is terrible on most, but mainly its staying connected to the ap's all the time and especially multiaccess points that causes issues. The hitachi phone I tried, the 5000, it was bad, it doesn't support wpa, that's crazy. No firmware updates in a while either so its not coming. The new one maybe does, the ae. The utstarcom one never stayed connected either. Anyways the best is what the other guy said, phones that are not wifi but integrated with sip, that might be worth looking into. I assume the hotel already has the access points that's why you are doing this? Well I can see the reason, my recommendation, do extensive testing first with the phones you are looking at, as in multi day testing to make sure the phones stay connected and get all the calls. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, November 25, 2007 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recommendations for 100 Wifi SIP phone setup Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight & Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for 100 Wifi SIP phone setup
On Sun, 25 Nov 2007 23:26:54 +0300, [EMAIL PROTECTED] wrote: >Hi all, > > >Im preparing a quote for a 5 Star hotel, planning to have around 100 >SIP Wifi phones for PBX operations running on 100 AccessPoints. >Network is running in ARUBA Networks - AP70 access points. > >The initial recommendation is to go for Hitachi Wifiphones, but i >would like to know from the group the recommendations. Im planning to >put up Asterisk as the PBX, Please advice me the do's and donts as i'm >not experienced on such heavy installation which are mission critical. >I had been using asterisk on small profiles and this would be my first >Pro setup with wifi handsets if all goes as planned. > >the Key Questions are > >Is Asterisk good enough? or do we need a another Proxy like SER? > >What is the experience with Hitachi Wifi phone's? Any specific Issues? > >Any such installations done? Please do a detail I had some Hitachi WIP5000 back in early 2006. It looks like a nice device but it really didn't deliver upon its promise. The reason to select a wifi phone is that by staying IP end to end you might gain operational advantages. It should have at least some of the features of a proper SIP deskset. The WIP5000 did not provide this at the time. The phones that I used had simple trouble moving between access points. Also the volume of the earpiece was very low, even for use in a quiet office. I am led to beleive that the new DECT cordless IP devices, like the system from Aastra Telecom, are currently a better option than wifi devices. Michael -- Michael Graves mgravesmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --Bandwidth and Colocation Provided 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] Recommendation for 100 SIP WiFi phone setup
Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
As SIP is not Analog FXO, my comments do not apply to them. I have no idea if or which analog adapters might detect line voltage or dialtone. Paul wrote: > Do the SIP-FXO gateway devices do any better? > > Eric "ManxPower" Wieling wrote: > >> Asterisk does not detect analog ports with no line plugged in. It does >> not test for dialtone before dialing (this applies to all analog cards >> except the X100P). >> >> Rilawich Ango wrote: >> >> >>> It works if it specified the port exactly plugged to PSTN. I want to >>> clarify the dial command here. >>> >>> Dial(zap/g1/1234567) >>> >>> It will try channel 1, if it is busy, congested then it will try >>> channel 2 and so on, right? >>> I wonder if I don't plug the PSTN to channel 1, there should not be a >>> dial tone on it. Why it still try channel 1 and make call using it? >>> >>> On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >>> >>> On Sat, 24 Nov 2007, Rilawich Ango wrote: > I have a TDM400 with all FXO module in it. Only one channel (say > channel 3) is plugged to PSTN. In my understand, a dial command > Dial(zap/g1/12345677) should search an available channel, which is 3, > in group 1 to make a call. However, I found that it will still use > channel 1 to make call even it hasn't plugged to the PSTN. Below are > the conf files. > > --zapata.conf-- > group=1 > signalling=fxs_ks > context=incoming > channel => 1-8 > > You really only want channel => 3 > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - 3Com and IBM iSeries
Olivier wrote: > Thanks for the tip but my question was more towards asking for return > of experience than asking technical capabilities. > > For a long time now, there's a lot of buzz surrounding server > consolidation > (http://searchdatacenter.techtarget.com/sDefinition/0,,sid80_gci1070272,00.html > > ) along "blade servers replacing multiple servers" but this is the > very first time I've met a customer seriously considering those things > for ToIP. > > And we all know the difference between touting "consolidation" and > meeting a customer ready to close a deal with it. > That's why I thought reporting this might be interresting. > > Seriously, would you bet a fully fledged Asterisk running an iSeries > LPARS ? With meetme and timing stuff ? > > Apparently, some are ready to bet on it (though I still have to meet > one running customer). > Thanks for the tips, anyway. > The IBM guy at Astricon and self proclaimed "Asterisk IBM Skunkworks guy" was to very convincing about Asterisk running on the iSeries except under Power Linux. I wish I remembered his name, but he would be a great guy to get all of your questions answered. He was not tasked with Asterisk and the iSeries, he went to his supervisors and was given the OK on time, resources, and a couple of guys. Thanks, Steve Totaro 888.777.1888 ___ --Bandwidth and Colocation Provided 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] Recommendations for 100 Wifi SIP phone setup
Hi all, Im preparing a quote for a 5 Star hotel, planning to have around 100 SIP Wifi phones for PBX operations running on 100 AccessPoints. Network is running in ARUBA Networks - AP70 access points. The initial recommendation is to go for Hitachi Wifiphones, but i would like to know from the group the recommendations. Im planning to put up Asterisk as the PBX, Please advice me the do's and donts as i'm not experienced on such heavy installation which are mission critical. I had been using asterisk on small profiles and this would be my first Pro setup with wifi handsets if all goes as planned. the Key Questions are Is Asterisk good enough? or do we need a another Proxy like SER? What is the experience with Hitachi Wifi phone's? Any specific Issues? Any such installations done? Please do a detail Looking for experiences.. Thanks Sunil Charly Manager - Business Planning KOLTELECOM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp as T.38 termination?
You can also take a look at the T.38 product from Attractel. http://attractel.com/faxterisk.php Disclaimer, I work for these guys. Chris On Sun, November 25, 2007 4:11 pm, Robert Moskowitz wrote: > Olivier wrote: > >> Robert, >> >> >> Do you mean T.38 passthrough ou T.38 to T.30 gateway ? >> The former is said to work with Asterisk 1.4 but the latter is not ... >> > I know about what Asterisk 1.4 can do. And Asterisk 1.2 only does T.30 > passthrough :) You need 'stuff' to handle fax. Stuff like spandsp, > IAXmodem and T38modem. > > > So using spandsp and rxtax and txfax you **SHOULD** be able to set up a > efax with T.38 systems. > > Also T38modem will make the connection to Hylafax for the same. > >> >> Cheers >> >> >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> >> asterisk-users mailing list To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > asterisk-users mailing list To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
Do the SIP-FXO gateway devices do any better? Eric "ManxPower" Wieling wrote: >Asterisk does not detect analog ports with no line plugged in. It does >not test for dialtone before dialing (this applies to all analog cards >except the X100P). > >Rilawich Ango wrote: > > >>It works if it specified the port exactly plugged to PSTN. I want to >>clarify the dial command here. >> >>Dial(zap/g1/1234567) >> >>It will try channel 1, if it is busy, congested then it will try >>channel 2 and so on, right? >>I wonder if I don't plug the PSTN to channel 1, there should not be a >>dial tone on it. Why it still try channel 1 and make call using it? >> >>On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> >> >>>On Sat, 24 Nov 2007, Rilawich Ango wrote: >>> >>> >>> I have a TDM400 with all FXO module in it. Only one channel (say channel 3) is plugged to PSTN. In my understand, a dial command Dial(zap/g1/12345677) should search an available channel, which is 3, in group 1 to make a call. However, I found that it will still use channel 1 to make call even it hasn't plugged to the PSTN. Below are the conf files. --zapata.conf-- group=1 signalling=fxs_ks context=incoming channel => 1-8 >>>You really only want >>> >>> channel => 3 >>> >>> ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
Asterisk does not detect analog ports with no line plugged in. It does not test for dialtone before dialing (this applies to all analog cards except the X100P). Rilawich Ango wrote: > It works if it specified the port exactly plugged to PSTN. I want to > clarify the dial command here. > > Dial(zap/g1/1234567) > > It will try channel 1, if it is busy, congested then it will try > channel 2 and so on, right? > I wonder if I don't plug the PSTN to channel 1, there should not be a > dial tone on it. Why it still try channel 1 and make call using it? > > On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> On Sat, 24 Nov 2007, Rilawich Ango wrote: >> >>> I have a TDM400 with all FXO module in it. Only one channel (say >>> channel 3) is plugged to PSTN. In my understand, a dial command >>> Dial(zap/g1/12345677) should search an available channel, which is 3, >>> in group 1 to make a call. However, I found that it will still use >>> channel 1 to make call even it hasn't plugged to the PSTN. Below are >>> the conf files. >>> >>> --zapata.conf-- >>> group=1 >>> signalling=fxs_ks >>> context=incoming >>> channel => 1-8 >> You really only want >> >>channel => 3 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp as T.38 termination?
Olivier wrote: > Robert, > > Do you mean T.38 passthrough ou T.38 to T.30 gateway ? > The former is said to work with Asterisk 1.4 but the latter is not ... I know about what Asterisk 1.4 can do. And Asterisk 1.2 only does T.30 passthrough :) You need 'stuff' to handle fax. Stuff like spandsp, IAXmodem and T38modem. So using spandsp and rxtax and txfax you **SHOULD** be able to set up a efax with T.38 systems. Also T38modem will make the connection to Hylafax for the same. > > Cheers > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 with LDAP
Hi, I think most of what you're looking for relates to "do you consider sip.confto describe users or resources" ? If it describes resources, how do you manage other IT resources (PC, printers, ...) ? Do you store devices passwords (BIOS passwords with Serial numbers, ...) in an LDAP database ? If it describes users, then obviously you would like to use the same tools as for other IT applications with LDAP databases and so on. >From my point of view, a good comparison would be to treat IP phones just like PCs : if you ask user to type a password to gain network access when arriving in the morning, then you could ask them to type a PIN code in the first time they are outcalling someone. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd bug in Siemens C460IP ?
Hi Paul, 2007/11/23, Paul Hayes <[EMAIL PROTECTED]>: > > Robert Lister wrote: > > Hello, > > > > I think I have encountered an odd bug in Siemens C460 IP/dect handsets, > > which is a bit annoying, and I'm not (yet) sure how to get round it > without > > lots of hacks. > > > > Basically, on all external incoming calls, we set: > > > > exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) > > > > This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a > > different ring cadence so to differentiate between external and internal > > calls. > > > > Other handsets that do not support Alert-Info: just ignore the presence > > of this header. > > > > When this header is set in a call to the C460 IP, it does not alert, in > fact > > it does not respond to any INVITE requests; asterisk just retries the > > requests a few times and then gives up. > > > > Anyone able to reproduce? I have firmware version 0107 / 041.00 > > > > I suppose as a workaround I could add an astDB entry for these > extensions, > > and a bit of logic in the dialplan to tell asterisk not to add the > header > > for extensions that have that flag set. > > > > > > Regards, > > > > > > > > Rob > > > > > > I can replicate this behaviour too using an S450IP when an Alert-Info > header is present. I have reported the issue to Siemens so hopefully > this will be fixed in a firmware update in the near future. > > cheers, > Paul. > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Could you get from Siemens some kind of commitment to fully support Alert-Info or at least, to ignore Alert-Info data in incoming INVITEs ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp as T.38 termination?
Robert, Do you mean T.38 passthrough ou T.38 to T.30 gateway ? The former is said to work with Asterisk 1.4 but the latter is not ... Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - 3Com and IBM iSeries
Thanks for the tip but my question was more towards asking for return of experience than asking technical capabilities. For a long time now, there's a lot of buzz surrounding server consolidation ( http://searchdatacenter.techtarget.com/sDefinition/0,,sid80_gci1070272,00.html) along "blade servers replacing multiple servers" but this is the very first time I've met a customer seriously considering those things for ToIP. And we all know the difference between touting "consolidation" and meeting a customer ready to close a deal with it. That's why I thought reporting this might be interresting. Seriously, would you bet a fully fledged Asterisk running an iSeries LPARS ? With meetme and timing stuff ? Apparently, some are ready to bet on it (though I still have to meet one running customer). Thanks for the tips, anyway. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial in group
On Sun, 25 Nov 2007, Rilawich Ango wrote: > It works if it specified the port exactly plugged to PSTN. I want to > clarify the dial command here. > > Dial(zap/g1/1234567) > > It will try channel 1, if it is busy, congested then it will try > channel 2 and so on, right? Yes. > I wonder if I don't plug the PSTN to channel 1, there should not be a > dial tone on it. Why it still try channel 1 and make call using it? Because asterisk can't tell if an analogue line is plugged in or not. To get a dial-tone, it would have to activate the line (ie. "lift the handset") and it's not going to do that. It's relying on the channel instruction in the zapata.conf file to tell it what lines are really live, so get them right and everything else will "just work". Gordon > On Nov 25, 2007 5:00 AM, Gordon Henderson <[EMAIL PROTECTED]> wrote: >> >> On Sat, 24 Nov 2007, Rilawich Ango wrote: >> >>> I have a TDM400 with all FXO module in it. Only one channel (say >>> channel 3) is plugged to PSTN. In my understand, a dial command >>> Dial(zap/g1/12345677) should search an available channel, which is 3, >>> in group 1 to make a call. However, I found that it will still use >>> channel 1 to make call even it hasn't plugged to the PSTN. Below are >>> the conf files. >>> >>> --zapata.conf-- >>> group=1 >>> signalling=fxs_ks >>> context=incoming >>> channel => 1-8 >> >> You really only want >> >>channel => 3 >> >> here if it's only channel 3 that's plugged in. >> >> Gordon >> >> ___ >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium and Asterisk
Michael J. Liberatore wrote: > Can you elaborate on OSLEC? I cant say I have heard of it but it sounds > very interesting considering it worked for x100p for you which was the > worst out of ALL the cards I have ever tried for echo. > > Thanks > > Mike > Hi Mike, Sure... OSLEC (Open Source Echo Canceller). Here's the hope page: http://www.rowetel.com/ucasterisk/oslec.html. On my blog I discussed installing it with zaptel 1.4.5.1. (1.4.6 is easier as you they fixed a bug in that needed manually patching.) http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-71-oslec/ I'd just try it out. If you read the thread on the Trixbox forum you'll see that most of the respondents had excellent results too. HTH Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
