Re: [asterisk-users] SOLVED: No reply to our critical packet
Hi, Next Step would be to check/update the firmware on your phones or router. I dismissed this advice at first, but it was the one that worked in the end. The D-Link DSL-2500U ADSL router was to blame, it must have been interfering with SIP packets (maybe an outdated version of the SIP conntrack module or something like that). The 1.50 firmware version solved the problem and also gave the impression of working faster overall than 1.20. Thanks a lot. Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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] BRI cards; JUNGHANNS AND B410P
Thank you for your attention; I have successfully installed junghanns (With BRIstuff) under a kernel 2.6 fc6 and asterisk 1.2, but i can't do it with B410P in the same environnement(Problem with the kernel);but my real problem is in the configuration of extentions.conf because i don't have a ISDN line to test it. THANKS A LOT. 2009/3/14 Olivier oza-4...@myamail.com 2009/3/14 Rayed Bs rayed.i...@gmail.com hi every body, can anyone give me the right configuration of BRI cards; zapata.conf , zaptel.conf ans extensions.conf; please help which config do you target ? b410p or junghanns ? which asterisk version ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] trying asterfax
Hi, Has anyone able to make asterfax work on asterisk, specifically asterisk 1.4. reading the documentation it prefers asterisk 1.2, unfortunately i've already setup my asterisk and it's working ok at that version. TIA. regards, nha ___ -- 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] t38 iax trunk
Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 - SIP ATA (T38 enabled) - fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys SPA2102, Asterisk 1.4.22 (configured with t38pt_udptl = yes) and I have a pretty good link so faxes are going through even if T38 is switched off. Interesting thing is that faxes are going through even when one ATA is T38 enabled and the other isn't... Thanks for help /dubravko ___ -- 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] BRI cards; JUNGHANNS AND B410P
On Mon, Mar 16, 2009 at 09:14:48AM +0100, Rayed Bs wrote: Thank you for your attention; I have successfully installed junghanns (With BRIstuff) under a kernel 2.6 fc6 and asterisk 1.2, but i can't do it with B410P in the same environnement(Problem with the kernel);but my real problem is in the configuration of extentions.conf because i don't have a ISDN line to test it. THANKS A LOT. AFAIK the B410P can work with a slightly modified version of qozap, but the hardware echo canceller will not be supported. I suppose that also implies that the LEDs won't work. There is a DAHDI driver for it as part of the standard DAHDI distribution. This means that you can use it with Asterisk 1.4 ( = 1.4.22) with bristuffed Asterisk and libpri and with Asterisk 1.6.x with standard Asterisk and libpri. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] url in dial command: how does it work?
Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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] SIMPLE
Hi All, Is this available on asterisk: http://www.ietf.org/html.charters/simple-charter.html what do i need to enable to support this. thanks Regards, Nhadie ___ -- 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] ANI with Pickup application
Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? Regards, Christophorus ___ -- 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] url in dial command: how does it work?
Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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 1.6 ReceiveFAX problem
MaxGao wrote: hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines. it seems the channel.c try to call ast_read(), read some bytes from the channel but there is nothing ... whether it's a loop to check data on the channel ? and many times when reciving tax , the E1 card will down , all the channel get red alarm... [Mar 16 09:49:19] DEBUG[20928] chan_dahdi.c: Monitor doohicky got event Alarm on channel 2 [Mar 16 09:49:19] WARNING[20928] chan_dahdi.c: Detected alarm on channel 2: Recovering i then try asterisk 1.4.23.2 and agx-ast-addon , when using spandsp 0.0.5 and spandsp 0.0.6, like above , sometimes all E1 channel get red alarm when reciving fax but use spandsp 0.0.4 get no error... some one can tell me what version of asterisk and spandsp is the best version for fax??? thanks a lot. Red alarms have nothing whatsoever to do with FAXing, or even Asterisk. They only relate to the state of the line, the card, and the dahdi/zaptel drivers. You'd better sort that out before expecting any application to work. Regards, Steve ___ -- 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] t38 iax trunk
dubravko caric wrote: Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 - SIP ATA (T38 enabled) - fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys SPA2102, Asterisk 1.4.22 (configured with t38pt_udptl = yes) and I have a pretty good link so faxes are going through even if T38 is switched off. Interesting thing is that faxes are going through even when one ATA is T38 enabled and the other isn't... There is no definition for how T.38 messages would be handled in an IAX stream, so I doubt you are sending T.38 data between those two Asterisk boxes. Regards, Steve ___ -- 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] ANI with Pickup application
2009/3/16 Christophorus Laube christophorus.la...@semanticedge.de Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? using SIP P-asserted ids and asterisk 1.6.1, this shoulld be possible to get CallerID (I've never tried it yet). Regards, Christophorus ___ -- 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] Transfers on an inter-PBX PRI link
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension
104
Step3 fails and extension 103 is reconnected to 101.
Why is Step3 failing and how could I change my setup so the transfer succeeds?
As a side question, I'd like to know if I could free the unnecessary zap
channels created in Steps 1 and 2.
On analog channels I could SendDTMF(${EXTEN}). I don't know how to do that on a
digital pri line and if it requires that the legacy PBX be compatible.
Anyway, I'm not too worried about freeing the PRI channels. I just want Step3
to work.
Is it possible, somehow?
Thanks in advance,
Vieri
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Re: [asterisk-users] url in dial command: how does it work?
Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Ignore switch to REVERSED Polarity on channel 1, state 4
Hi, Trying to trace an asterisk hang on a production (it had to be didn't it) system. The last thing before it crashed was [Mar 16 12:32:42] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 4 [Mar 16 12:54:34] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 12:54:35] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 12:55:09] DEBUG[7754] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 Whilst i am aware polarity reversal is normal (clid stuff etc) the 'state 4' is not... see this grep for 'REVERSED' in full log: [Mar 16 08:01:22] DEBUG[11561] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:05:02] DEBUG[11576] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:43:43] DEBUG[11737] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:45:16] DEBUG[11747] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:50:22] DEBUG[11767] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:51:02] DEBUG[11771] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:51:40] DEBUG[11776] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:52:05] DEBUG[11780] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 [Mar 16 08:53:41] DEBUG[11786] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:58:14] DEBUG[11798] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 08:59:55] DEBUG[11803] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:11:49] DEBUG[11828] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:14:23] DEBUG[11836] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:20:47] DEBUG[11851] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:26:08] DEBUG[11949] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:26:48] DEBUG[11958] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 [Mar 16 09:27:34] DEBUG[11968] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:39:19] DEBUG[11999] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:45:22] DEBUG[12015] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:48:59] DEBUG[12033] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 09:49:35] DEBUG[12038] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:52:00] DEBUG[12046] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 [Mar 16 09:52:02] DEBUG[12045] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:53:37] DEBUG[12054] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:53:57] DEBUG[12058] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 [Mar 16 09:55:50] DEBUG[12073] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:56:01] DEBUG[12076] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 6 [Mar 16 09:57:26] DEBUG[12081] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 09:58:21] DEBUG[12086] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:04:59] DEBUG[12101] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:12:43] DEBUG[12117] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:17:34] DEBUG[12141] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:17:46] DEBUG[12145] chan_zap.c: Ignore switch to REVERSED Polarity on channel 3, state 6 [Mar 16 10:18:27] DEBUG[12150] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:21:42] DEBUG[12164] chan_zap.c: Ignore switch to REVERSED Polarity on channel 2, state 4 [Mar 16 10:23:10] DEBUG[12172] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:39:19] DEBUG[12282] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:45:24] DEBUG[12298] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:46:15] DEBUG[12303] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:50:24] DEBUG[12316] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:57:21] DEBUG[12331] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:58:57] DEBUG[12338] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Mar 16 10:59:43] DEBUG[12342] chan_zap.c: Ignore
Re: [asterisk-users] url in dial command: how does it work?
Hi Tim, it seems that using trunks is the right wayis this what you meant? Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 -- _ Giorgio Incantalupo, mailto:gincantal...@fgasoftware.com FGA srl - http://www.fgasoftware.com - vo...@work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Transfers on an inter-PBX PRI link
On Mon, Mar 16, 2009 at 8:49 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension
104
Step3 fails and extension 103 is reconnected to 101.
Why is Step3 failing and how could I change my setup so the transfer succeeds?
As a side question, I'd like to know if I could free the unnecessary zap
channels created in Steps 1 and 2.
On analog channels I could SendDTMF(${EXTEN}). I don't know how to do that on
a digital pri line and if it requires that the legacy PBX be compatible.
Anyway, I'm not too worried about freeing the PRI channels. I just want
Step3 to work.
Is it possible, somehow?
Thanks in advance,
Vieri
Relevant parts of your dialplan, tech.conf, and debug info is probably
the only way to really help you besides making wild guesses.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] ANI with Pickup application
Hallo Ralf, das ist die Antwort von der Liste. Klingt etwas vage und nicht absolut erfolgversprechend... Derzeit ist kein Upgrade auf Asterisk 1.6 geplant und nach allem, was ich darüber bisher gelesen habe, kann die Umstellung auch etwas größer werden (Wählplansyntax etc.). Beronet empfiehlt derzeit auch nach wie vor 1.4, von 1.6 auf Produktivsystemen wird abgeraten. Gruß, Christophorus 2009/3/16 Christophorus Laube christophorus.la...@semanticedge.de Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? using SIP P-asserted ids and asterisk 1.6.1, this shoulld be possible to get CallerID (I've never tried it yet). Regards, Christophorus ___ -- 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 -- Dipl.-Ling. Christophorus Laube Systemadministrator SemanticEdge GmbH Kaiserin-Augusta-Allee 10-11 10553 Berlin Deutschland Tel +49-30-345077-58 Fax +49-30-345077-77 christophorus.la...@semanticedge.de Geschäftsführer : Dr.Ralf Köhrbrück, Dr. Lupo Pape HRB 84682 ___ -- 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] Problems on default Attended Transfer
Hi,
I'm currently using Asterisk 1.4.23.1, and I have a problem (also on
previous version).
Sometimes, when I try to do an attended transfer to another internal with
default feature *2, Asterisk doesn't make it (it doesn't play
'pbx-transfer'). Sometimes on second time, Asterisk make transfer correctly.
I have this problem on variuos type of SIP phones (GrandStream, Aastra,
OKI).
My sip.conf is like the following account:
===
[intphones](!)
type=friend
qualify=yes
host=dynamic
callgroup=1
pickupgroup=1
dtmfmode=sip
[1](intphones)
context=IntPhones
username=1
secret=1234
amaflags=documentation
accountcode=11
subscribecontext=IntPhones
callerid=phone 11 11
limitonpeers=yes
call-limit=100
[2](intphones)
context=IntPhones
username=2
secret=1234
amaflags=documentation
accountcode=12
subscribecontext=IntPhones
callerid=phone 12 12
limitonpeers=yes
call-limit=100
===
and on extensions.conf my dial lines are like:
===
exten = _1X,1,Dial(SIP/${EXTEN:1},,tTr)
exten = _1X,n,Hangup()
===
Can anyone help me? I don't underwstand where I make the mistake!
Thanks to everyone
Marco
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Re: [asterisk-users] Are .call files working with extensions.ael ? bristuff problem
Hi,
As soon as I removed back line 266 as suggested by Peer Oliver, it worked.
Lines changed in /usr/src/bristuff-0.4.0-RC4-xr6/asterisk/pbx/pbx_spool.c :
/* Olivier
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten)) ||
(ast_strlen_zero(o-message) ast_strlen_zero(o-pdu))) {
*/
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten))) {
Procedure used to update:
cd /usr/src/bristuff-0.4.0-RC4-xr6/asterisk/
make all
make install
File astup.call :
Channel: Sip/700
Context: mylocal
Extension: 00123457530
Priority: 1
Now, looking at removed line, it would say if both message and pdu are
empty, then print error message.
Question is now, is this a feature (ie you must either add a pdu or a
message (or both) in call files) or a bug ?
Regards
PS: I'll post the answer to Bristuff mailinglist as this must of interest
there ...
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Re: [asterisk-users] url in dial command: how does it work?
Oh sorry, I wasn't clear. The IAX protocol has a frame type for sending this URL info. Skype has an attribute for it. The intention is (I think) to be able to forward the URL for the customer (in the corporate CRM system) to the agent answering a call on a softphone. Some of the IAX softphones support this. What were you planning to do with it. Tim. On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote: Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] 428 Loop Detected
On Mon, Mar 16, 2009 at 12:10 AM, Steve Totaro stot...@totarotechnologies.com wrote: Again, if I am interpreting this correctly, he is not using SIP. A four port card 2fxo/2fxs means to me that he is not using SIP at all. You are correct. I was confused. It is Zap (zaptel) channel If by card, you mean some kind of SIP gateway, then I misunderstood and the problem, but seeing DAHDI channels leads me to believe that SIP is not required and actually causing your problems. SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this case)... If you had a SIP device, it would be connected to the data network, not a phone line. Can you just plug your phone into a regular landline jack and get dialtone? If so, forget SIP for now. Comment out or delete all your sip.conf peers since you are not using SIP. Change your dialplan to not (Dial/SIP but (Dial/DAHDI/1,10) and the correct channel to your FXS port that the phone is connected to. Dial(Zap/1) worked like a charm. Thanks all for your help Thanks, Steve Totaro On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta marco.mo...@gmail.com wrote: Hi, problem is that you are saying that phone in sip.conf is at the same ip address of your asterisk box so you are dialing into a loop to your self asterisk box [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 what you need is: [phone] type=friend context=phone1 secret=g00dpazzwerd dtmfmode=rfc2833 host=dynamic ;configuring your codecs (i don't know what else you have configured, just preventing audio for you) disallow=all allow=ulaw allow=alaw allow=gsm Dial sip/phone is enough too.. [from-pstn] ;include = default exten = s,1,Dial(SIP/phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup hope it helps. don't forget to asterisk reload on cli. Looking forward to hearing from you. cheers -- Marco Mouta On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote: Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqb...@improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106 dtmfmode=rfc2833 iqb...@improvise:/etc/asterisk$ cat extensions.conf [default] exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Playback(tt-monkeys) exten = s,4,Hangup [from-internal] include = default [phone1] [from-pstn] ;include = default exten = s,1,Dial(SIP/ph...@phone,10) exten = s,2,Voicemail(line) exten = s,3,Hangup [line1] So my home land line is going to the FXO port and my home phone is hanging off of FXS port. Here are the contexts for my fxo/fxs card improvise*CLI dahdi show channels Chan Extension Context Language MOH Interpret pseudo default default 1 from-internal default 2 from-internal default 3 from-pstn default 4 from-pstn default I want to call from my cell and make my home phone ring and if I dont pickup in 10 secs I want the call go to my voicemail. But I am getting a loop detect. The debug output is attached. What am I doing wrong? -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? ___ -- 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] t38 iax trunk
On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
dubravko_ca...@yahoo.com wrote:
fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 -
SIP ATA (T38 enabled) - fax
My question is, how can I know if I'm really using T38? is T38 information
coming to the other side (because of SIP to IAX conversion) or just plain
g711a data?
You don't give dialplan samples, but if you're using Asterisk SendFax
and ReceiveFax from app_fax...
You can answer this directly in your dialplan:
exten = s,1,Answer
exten = s,n,Set(LOCALSTATIONID=faxmodem01)
exten = s,n,Set(LOCALPATH=/var/spool/fax/recvq/${LOCALSTATIONID}/)
exten = s,n,Set(MYLOCALDATE=${STRFTIME(${EPOCH},,%C%y-%m-%d-%H%M)})
exten = s,n,Set(MYFILENAME=${LOCALSTATIONID}-${MYLOCALDATE}-${CDR(uniqueid)})
exten = s,n,Set(MYFULLPATH=${LOCALPATH}${MYFILENAME})
exten = s,n,ReceiveFax(${MYFULLPATH}.tif)
exten = h,1,System(/bin/echo
${MYFULLPATH},${CALLERID(num)},${CALLERID(name)},${FAXSTATUS},${FAXERROR},${FAXMODE},${FAXPAGES},${REMOTESTATIONID}
${LOCALPATH}fax.log)
exten = h,n,Hangup
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[asterisk-users] A400P + Intel D201GLY2(A) motherboard?
Hello I'd like to build myself an Asterisk server for SOHO use. Intel's D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very good deal, but I'm concerned about two things: 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible http://tinyurl.com/ck6nfu) fit with a PCI riser, or will the CPU heatsink/fan be in the way? I downloaded the PDF from OpenVox's site, but didn't find the size of each PCI card 2. Performance, especially if there's the need for software echo cancelling If someone here has used this motherboard to build an Asterisk server, could you answer the questions, at least the first one? 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] No hardware timing source found in /proc/dahdi
John Millican wrote:
Well,
lsmod | grep hisax returns nothing
plain lsmod:
Module Size Used by
dahdi_dummy22472 0
dahdi 215776 1 dahdi_dummy
crc_ccitt 18944 1 dahdi
af_packet 57100 2
snd_pcm_oss67456 0
snd_mixer_oss 34176 1 snd_pcm_oss
snd_seq74992 0
snd_seq_device 25620 1 snd_seq
vmnet 72992 3
parport_pc 58456 0
parport56588 1 parport_pc
vmmon 158908 0
sunrpc198600 1
iptable_filter 19840 0
ip_tables 37848 1 iptable_filter
ip6table_filter19584 0
ip6_tables 31944 1 ip6table_filter
x_tables 37000 2 ip_tables,ip6_tables
ipv6 372344 29
cpufreq_conservative24968 0
cpufreq_userspace 23680 0
cpufreq_powersave 18560 0
powernow_k831504 0
apparmor 58672 0
loop 36356 0
dm_mod 77152 0
ohci1394 51272 0
ieee1394 115800 1 ohci1394
i2c_nforce222784 0
snd_hda_intel 368804 0
i2c_core 43648 1 i2c_nforce2
snd_pcm 108680 2 snd_pcm_oss,snd_hda_intel
snd_timer 42632 2 snd_seq,snd_pcm
snd84984 7
snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer
k8temp 22656 0
hwmon 20232 1 k8temp
button 26400 0
usblp 30976 0
forcedeth 65416 0
rtc_cmos 25016 0
rtc_core 38156 1 rtc_cmos
rtc_lib19968 1 rtc_core
sr_mod 33444 0
cdrom 52392 1 sr_mod
usb_storage 102816 0
soundcore 25360 1 snd
snd_page_alloc 27280 2 snd_hda_intel,snd_pcm
ide_core 165648 1 usb_storage
sg 53304 0
usbhid 58160 0
hid43776 1 usbhid
ff_memless 22536 1 usbhid
sd_mod 45824 6
ohci_hcd 38020 0
ehci_hcd 50572 0
usbcore 155560 6 usblp,usb_storage,usbhid,ohci_hcd,ehci_hcd
edd26760 0
ext3 156688 3
mbcache26248 1 ext3
jbd89192 1 ext3
fan22792 0
sata_nv38404 4
pata_amd 31876 0
libata164096 2 sata_nv,pata_amd
scsi_mod 176536 5 sr_mod,usb_storage,sg,sd_mod,libata
thermal34576 0
processor 59592 2 powernow_k8,thermal
Looking at the lsmod output, it appears that the wctdm module is not
loaded. So either the /etc/dahdi/modules has the wctdm module commented
out, or something is wrong with the /etc/init.d/dahdi that it isn't
viewing that file.
If you unload all the drivers ('/etc/init.d/dahdi stop') and make sure
they are unloaded ('lsmod | grep dahdi' should not show any output) then
just load the wctdm driver ('modprobe wctdm'), and then what does dmesg
show?
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Ast/Hyla/IAX Scalability?
David Backeberg wrote: On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood ste...@coppice.org wrote: Fully open-to-the-public FAX servers tend to get just get a lot of bad calls, many of them wrong numbers, or voice users. FAX servers for I've definitely seen that, and have been able to either identify the validity of a caller by CID or by calling the number and confirming a blast of fax tones. clue what kind of failure rate might be expected. You can find a bit more about these issues and our results at http://www.soft-switch.org/spandsp-soft-fax-performance.html After reading that, it occurred to me that I'm running SpanDSP 0.0.5 and 0.0.6 seems to have enhancements that may solve the problems I've been seeing. I'm convinced that it's worth upgrading and seeing if I can reduce my failure rate. Your differing failure rates between using ReceiveFAX and using iaxmodem seem to indicate your results relate to issues in your own system, I think I wasn't very good at setting it up, as I had no experience with IAX. Likely my fault rather than anything inherently wrong with the software. There were more moving parts than I was able to get a handle on, and when I switched to 1.6 and app_fax things 'just worked'. This is why I keep recommending the 1.6 approach over the 1.4 + IAX + IAXModem + Hylafax. LANs don't loose packets), will have a true failure rate (i.e. a rate of calls failing which had the potential to succeed) well below 1%. The That's consistent with my testing before I set it live. You mentioned recording faxes. I know how to do that with IAXModem, but are you familiar with a method for 1.6 and app_fax? I read through app_fax.c and didn't see any way to send a flag. Is the recording built into SpanDSP, or is is something IAXModem added on themselves? For what it's worth, the company I work for switched from WinFax to HylaFax last spring. We only have 4 analog phone lines coming in to a 4-port modem card, but the Hylafax system runs on the same server as our main Asterisk PBX. So far Hylafax is performing much better than WinFax ever did. When we have errors either sending or receiving, it is always either line problems or the wrong number being dialed resulting in a voice call to the fax line. I would estimate that our overall success rate is around 95% if you disregard faxes to wrong numbers or incoming voice calls to the fax lines. Load testing a large-scale fax system under real-world conditions is difficult if not impossible without having access to a variety of hardware and software fax devices scattered all over your prospective send or receive area. If you load test from your own location by attaching a bunch of fax machines or a fax sending server to your outgoing lines and have them dial back in, then you're only looping through your local telco's switching center. You might get very different results from sending faxes from out of state, or even across town. It's been my experience that telephone line quality varies greatly from place to place and even from time to time. A perfect example is from back in my days as a systems admin for a dial-up ISP. We were operating in a small town where PRI or channelized T1's weren't available so we had a bank of about 100 US Robotics external modems connected with serial cables to 2 Livingston PortMaster terminal servers. Everything would run fine (or as fine as it ever got with dial-up) until it decided to rain. Everytime we'd get more than a tenth of an inch of rain a large group of the modems would go haywire and start dropping calls. A couple of the modems would burn out completely. We had the telco out repeatedly and they always gave us some answer that didn't make any sense. After about the 6th time this happened they sent out a technician with a brand new line analyzer that happened to include a TDR. The vast majority of the lines we were having trouble with showed to have a partial short about 100 feet from our building which just happened to be right under the middle of the road in front of our building. They dug the section of line up and found that the cable had been partially cut at some point in the past and the wires were spliced with electrical tape and the whole bundle had then been wrapped with tape. Every time it rained, the water would seep into the shoddy splice and short all the lines together. When the water dried out, the shorts would go away and the lines would go back to normal. I've seen situation like that enough to know that until everybody has a purely digital phone line, there will always be line quality problems that will be out of the end user's control. Even though the company I work for now is a small company is a very rural area where technology is somewhat limited, we're beginning to realize just how antiquated Fax is becoming. E-mail and web services are rapidly replacing fax to the point that 90% of
Re: [asterisk-users] t38 iax trunk
On Mon, Mar 16, 2009 at 11:29 AM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Mar 16, 2009 at 6:05 AM, dubravko caric
dubravko_ca...@yahoo.com wrote:
fax - SIP ATA (T38 enabled) - Asterisk #1 - IAX TRUNK - Asterisk #2 -
SIP ATA (T38 enabled) - fax
My question is, how can I know if I'm really using T38? is T38 information
coming to the other side (because of SIP to IAX conversion) or just plain
g711a data?
${MYFULLPATH},${CALLERID(num)},${CALLERID(name)},${FAXSTATUS},${FAXERROR},${FAXMODE},${FAXPAGES},${REMOTESTATIONID}
${LOCALPATH}fax.log)
exten = h,n,Hangup
Once you put that in place, the FAXMODE variable will tell you whether
the fax came through as T38 or voice.
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Re: [asterisk-users] A400P + Intel D201GLY2(A) motherboard?
On Mon, 16 Mar 2009, Gilles wrote: Hello I'd like to build myself an Asterisk server for SOHO use. Intel's D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very good deal, but I'm concerned about two things: 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible http://tinyurl.com/ck6nfu) fit with a PCI riser, or will the CPU heatsink/fan be in the way? I downloaded the PDF from OpenVox's site, but didn't find the size of each PCI card It depends on the riser and the case you use - if it extends out away from the board, you'll not have any issues, (pizza box type case) but if it doubles-back over the board (shuttle/cube type case) then there may not be enough headroom. (The fan/heatsink looks high, but can't really tell from the photos) 2. Performance, especially if there's the need for software echo cancelling Zero issues with it's performance for SOHO use IMO. I don't use that processor/board, but do use 1GHz VIA processors. Oslec works great on openvox cards. 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
Re: [asterisk-users] A400P + Intel D201GLY2(A) motherboard?
Gilles wrote: Hello I'd like to build myself an Asterisk server for SOHO use. Intel's D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very good deal, but I'm concerned about two things: 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible http://tinyurl.com/ck6nfu) fit with a PCI riser, or will the CPU heatsink/fan be in the way? I downloaded the PDF from OpenVox's site, but didn't find the size of each PCI card The card goes the other way, it doesn't go on top of the board. Well, at least there are risers going away from the board, I don't know if there are any going on top of it. 2. Performance, especially if there's the need for software echo cancelling I did some tests on it, not many. Without going higher than 2.0 load average I managed to do 10 calls per second, lasting 5 seconds each. During those 5 seconds, 2 sound files were played (sln). MySQL CDR was enabled, so that's also 10 DB writes/second. I don't know exactly what board it was, but the processor was a Atom =2,2GHz. It had fan. Two cards were used at the same time, one B400P and one A800, both Openvox. ___ -- 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] Busy on SIP
Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes host=dynamic callgroup=0 pickupgroup=0 context=office username=10 secret=1234 subscribecontext=BLF_group limitonpeers=yes call-limit=1 notifyringing=yes dtmfmode=info Someone can help me? I can't understand why I find it avaible when it makes an outgoing call. Thanks all Marco ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No hardware timing source found in /proc/dahdi
Shaun Ruffell wrote:
John Millican wrote:
Well,
lsmod | grep hisax returns nothing
plain lsmod:
Module Size Used by
dahdi_dummy22472 0
dahdi 215776 1 dahdi_dummy
crc_ccitt 18944 1 dahdi
af_packet 57100 2
snd_pcm_oss67456 0
snd_mixer_oss 34176 1 snd_pcm_oss
snd_seq74992 0
snd_seq_device 25620 1 snd_seq
vmnet 72992 3
parport_pc 58456 0
parport56588 1 parport_pc
vmmon 158908 0
sunrpc198600 1
iptable_filter 19840 0
ip_tables 37848 1 iptable_filter
ip6table_filter19584 0
ip6_tables 31944 1 ip6table_filter
x_tables 37000 2 ip_tables,ip6_tables
ipv6 372344 29
cpufreq_conservative24968 0
cpufreq_userspace 23680 0
cpufreq_powersave 18560 0
powernow_k831504 0
apparmor 58672 0
loop 36356 0
dm_mod 77152 0
ohci1394 51272 0
ieee1394 115800 1 ohci1394
i2c_nforce222784 0
snd_hda_intel 368804 0
i2c_core 43648 1 i2c_nforce2
snd_pcm 108680 2 snd_pcm_oss,snd_hda_intel
snd_timer 42632 2 snd_seq,snd_pcm
snd84984 7
snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_intel,snd_pcm,snd_timer
k8temp 22656 0
hwmon 20232 1 k8temp
button 26400 0
usblp 30976 0
forcedeth 65416 0
rtc_cmos 25016 0
rtc_core 38156 1 rtc_cmos
rtc_lib19968 1 rtc_core
sr_mod 33444 0
cdrom 52392 1 sr_mod
usb_storage 102816 0
soundcore 25360 1 snd
snd_page_alloc 27280 2 snd_hda_intel,snd_pcm
ide_core 165648 1 usb_storage
sg 53304 0
usbhid 58160 0
hid43776 1 usbhid
ff_memless 22536 1 usbhid
sd_mod 45824 6
ohci_hcd 38020 0
ehci_hcd 50572 0
usbcore 155560 6 usblp,usb_storage,usbhid,ohci_hcd,ehci_hcd
edd26760 0
ext3 156688 3
mbcache26248 1 ext3
jbd89192 1 ext3
fan22792 0
sata_nv38404 4
pata_amd 31876 0
libata164096 2 sata_nv,pata_amd
scsi_mod 176536 5 sr_mod,usb_storage,sg,sd_mod,libata
thermal34576 0
processor 59592 2 powernow_k8,thermal
Looking at the lsmod output, it appears that the wctdm module is not
loaded. So either the /etc/dahdi/modules has the wctdm module commented
out, or something is wrong with the /etc/init.d/dahdi that it isn't
viewing that file.
If you unload all the drivers ('/etc/init.d/dahdi stop') and make sure
they are unloaded ('lsmod | grep dahdi' should not show any output) then
just load the wctdm driver ('modprobe wctdm'), and then what does dmesg
show?
Well that did it. I guess I will have to modify /etc/init.d/dahdi to
only modprobe wctdm for now and run with it. wctdm was the only module
that was not commented out in /etc/dahdi/modules so it must be as you
said the /etc/init.d/dahdi was not reading the file as ity should. I
will look into what is happening there.
dmessg output:
dahdi: Telephony Interface Unloaded
dahdi: Telephony Interface Registered on major 196
dahdi: Version: 2.1.0.4
Freshmaker version: 71
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
dahdi_echocan_mg2: Registered echo canceler 'MG2'
dahdi: Registered tone zone 0 (United States / North America)
dahdi_cfg output:
DAHDI Tools Version - 2.1.0.2
DAHDI Version: 2.1.0.4
Echo Canceller(s): MG2
Configuration
==
Channel map:
Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
4 channels to configure.
Changing signalling on channel 1 from Unused to FXO Kewlstart
Setting echocan for channel 1 to mg2
Changing signalling on channel 2 from Unused to FXO Kewlstart
Setting echocan for channel 2 to mg2
Changing signalling on channel 3 from Unused to FXS Kewlstart
Setting echocan for channel 3 to mg2
Changing signalling on channel 4 from Unused to FXS Kewlstart
Setting echocan for channel 4 to mg2
Thank you very much for your
Re: [asterisk-users] Busy on SIP
On Mon, 16 Mar 2009, Marco Sambo wrote: Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. Disable call-waiting inside the phone. 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
Re: [asterisk-users] A400P + Intel D201GLY2(A) motherboard?
Paulo Santos wrote: I managed to do 10 calls per second, lasting 5 seconds each. 10 or 5, I can't remember... ___ -- 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] Aastra 9133i programmable buttons (* 4.1.23)
Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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
[asterisk-users] SIP audio delay after call transfer?
I have a customer with an Asterisk 1.4 system (r144238 - between 1.4.22-rc5 and 1.4.22 released). It uses SIP to connect to the PSTN via a provider who is on the same LAN as the box (it is co-located at the provider). They also have about 20 SIP phones as extensions that connect to the box over the internet. sip show peers indicates that most phones have a latency of 90ms-100ms. The provider is at 1ms. All links use the digium G.729 codec. They have reported that while call quality is normally very good, if a call is transferred from one extension to another, the transferred call starts to experience considerable audio latency. Transferring the call again also increases this latency even more, such that the call is unusable. My suspicion is that while performing the transfer, audio frames are building up somewhere and not being flushed (lack of autoservice somewhere in the code?). Has anyone else observed this behaviour? Even better, has anyone got a fix, or knows of such an issue having been fixed in a later version? This is a production system, so I can't easily try different versions to experiment, but could justify the downtime to install a known solution. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
2009/3/16 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Marco Sambo wrote: Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. Disable call-waiting inside the phone. Doesn't call-limit=1 force the same behaviour ? 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
Re: [asterisk-users] url in dial command: how does it work?
Hi Tim, I've made a test with 2 Asterisks and the 2 consoles showed me an HTML packet sent and one received. This does not work with the SIP protocol. The idea was to understand what was it for (I suppose someone did it for some purpose...), then how to use it to improve our solution (es: open pop ups) but we use SIP phones which do not support that URL parameter. I know queuemetrics use it but I cannot undestand how since tha URL parameter is passed to the called party while queuemetrics reads the queues.log file. BTW thanks for your time. Giorgio Tim Panton wrote: Oh sorry, I wasn't clear. The IAX protocol has a frame type for sending this URL info. Skype has an attribute for it. The intention is (I think) to be able to forward the URL for the customer (in the corporate CRM system) to the agent answering a call on a softphone. Some of the IAX softphones support this. What were you planning to do with it. Tim. On 16 Mar 2009, at 13:04, Giorgio Incantalupo wrote: Hi Tim, ok, but I think the big question is...what is the URL for? It seems I need a special device...but which? What kind of device do you use? Thanks. Giorgio Tim Panton wrote: Use IAX :-) In principle chan_skype could also support it. T. On 16 Mar 2009, at 10:51, Giorgio Incantalupo wrote: Hi, Does anybody knows where I can find some docs about how to make the URL parameter inside the Dial command work? I tried to make some tests with a sip phone without success: the sip debug shows no URL inside sip packets. :( Any hint appreciated. :) Thank you Giorgio ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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] Could Asterisk be rewriting an incoming invite?
I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: sip:216.82.224.202;lr;ftag=VPSF506071629460. Record-Route: sip:4.79.212.229;lr;ftag=VPSF506071629460. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0. Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501. From: BANDWIDTH COM sip:+19192282...@4.68.250.148sip%3a%2b19192282...@4.68.250.148 ;tag=VPSF506071629460. To: sip:+15129616...@4.79.212.229:5060. Call-ID: houmgc0520090316161653037...@209.244.63.35. CSeq: 1 INVITE. Contact: sip:+19192282...@4.68.250.148:5060;transport=udp. Max-Forwards: 67. Content-Type: application/sdp. Content-Length: 177. Remote-Party-ID: BANDWIDTH COM sip:+19192282...@4.68.250.148sip%3a%2b19192282...@4.68.250.148;party=calling ;screen=no;privacy=off. . v=0. o=- 1237220213 1237220214 IN IP4 209.244.187.176. s=-. c=IN IP4 209.244.187.176. t=0 0. m=audio 60458 RTP/AVP 0 18 101. a=rtpmap:101 telephone-event/8000. but asterisk is reporting it like this: INVITE sip:+15129616...@216.82.224.202:5060;transport=udp SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=VPSF506071629460 Record-Route: sip:4.79.212.229;lr;ftag=VPSF506071629460 Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0 Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK6314.15486fb6.0 Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207516720501 From: BANDWIDTH COM sip:+19192282...@4.68.250.148sip%3a%2b19192282...@4.68.250.148 ;tag=VPSF506071629460 To: sip:+15129616...@4.79.212.229:5060 Call-ID: houmgc0520090316161653037...@209.244.63.35 CSeq: 1 INVITE Contact: sip:+19192282...@4.68.250.148:5060;transport=udp Max-Forwards: 67 Content-Type: application/sdp Content-Length: 175 Remote-Party-ID: BANDWIDTH COM sip:+19192282...@4.68.250.148sip%3a%2b19192282...@4.68.250.148 ;party=calling;screen=no;privacy=off v=0 o=- 1237220213 1237220214 IN IP4 216.82.224.202 s=- c=IN IP4 216.82.224.202 t=0 0 m=audio 60458 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 as a result, I don't get incoming audio for obvious reasons. Is there any possibility that it's my asterisk configuration? I'm having a bear of a time getting to someone useful at my ISP, so I'm hoping to find that it's my problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra 9133i programmable buttons (* 4.1.23)
2009/3/16 David Ruggles da...@safedatausa.com: Is it possible to control the light on a programmable button without the blf option? I'm using a programmable button to turn call recording on and off and I would like the light to indicate the status. Thanks, 9133i phones are pretty much obsolete, and are not getting firmware updates, so I do not know whether Aastra ever put any of their XML application control code into that model. If they did, then it should be possible to respond with button status using XML updates from the server, otherwise you'd need to upgrade to one of their currently supported phones, which are almost certainly capable of this sort of thing. PS. I have never personally used the XML facility of Aastra phones, but I hear quite good things about it. Regards, Steve ___ -- 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] Could Asterisk be rewriting an incoming invite?
I've just determined that it IS happening on my box, but why? I did a packet capture using tcpdump on this very same box and it shows the correct invite while sip debug shows the wrong values. here's what I see in wireshark: No. TimeSourceDestination Protocol Info 1 0.00216.82.224.20267.198.16.18 SIP/SDP Request: INVITE sip:+15129616...@67.198.16.18:5060;transport=udp, with session description Frame 1 (1043 bytes on wire, 1043 bytes captured) Ethernet II, Src: EciTelec_00:a0:41 (00:02:0e:00:a0:41), Dst: Intel_92:3b:be (00:0c:f1:92:3b:be) Internet Protocol, Src: 216.82.224.202 (216.82.224.202), Dst: 67.198.16.18 (67.198.16.18) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0 Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 1237225281 1237225282 IN IP4 209.244.187.171 Session Name (s): - Connection Information (c): IN IP4 209.244.187.171 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 60570 RTP/AVP 0 18 101 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 and here's what I see in sip debug: INVITE sip:+15129616...@67.198.16.18:5060;transport=udp SIP/2.0 Record-Route: sip:216.82.224.202;lr;ftag=VPSF506071629460 Record-Route: sip:4.79.212.229;lr;ftag=VPSF506071629460 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK525.4ab0348.0 Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bK525.3b6e7ab3.0 Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207517079314 From: GARRIGUES,CHRIS sip:+15124990...@4.68.250.148sip%3a%2b15124990...@4.68.250.148 ;isup-oli=0;tag=VPSF506071629460 To: sip:+15129616...@4.79.212.229:5060 Call-ID: houmgc0520090316174121064...@209.244.63.35 CSeq: 1 INVITE Contact: sip:+15124990...@216.82.224.202:5060;transport=udp Max-Forwards: 67 Content-Type: application/sdp Content-Length: 175 Remote-Party-ID: GARRIGUES,CHRIS sip:+15124990...@4.68.250.148sip%3a%2b15124990...@4.68.250.148 ;party=calling;screen=yes;privacy=off v=0 o=- 1237225281 1237225282 IN IP4 216.82.224.202 s=- c=IN IP4 216.82.224.202 t=0 0 m=audio 60570 RTP/AVP 0 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 What is rewriting my o= and c= ?? ___ -- 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] Help Inbound number
i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Help Inbound number
Do you have an extension set for 246463 in your extensions.conf? On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.comwrote: i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] asterisk and ericsson e1 connection how to??
Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci::04:08.0 wcte12xp-d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But when i try to create a dahdi trunk, and dial it from asterisk , no call can be made. and also, when i try to call from ericsson side, i get line busy message as soon as i dial the number. Is there any guide that can help me in installing that card? PS: Whatever i made in SPAN config, everytime the only thing i see was Internal clock on dahdi_tool . How can i make my e1 card master (or slave whatever) instead of internal clock?? and other thing i wonder, if i create a span like span=1,0,0,ccs,hdb3 is it zap/g1 in zaptel(dahdi) conf menu in asteriskgui???(or freepbx) ___ -- 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] Contact id protocol problem
Hi, I'm using an Asterisk box with zap channel as a gateway between PSTN and an alarm receiver system. The alarm system uses Contact ID protocol. My problem is that the negotiation fails and I think that the problem is that kissoff tone is cut and the transmitter doesn't recognize it. Maybe the asterisk tone duration isn't long enough. I'm thinking about increasing the toneduration value in zapata.conf. or changind DTMF tone frecuency. Does anyone deal with a similar problem? What are the optimal values? Thanks 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] Help Inbound number
in my extension.conf i set : [default] exten = 1246463,1,Answer(SIP/8003) On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno tipas...@gmail.com wrote: Do you have an extension set for 246463 in your extensions.conf? On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 and ericsson e1 connection how to??
You should be able to get support from the people who sold you the card. You need to configure 2 files (I'm looking at an old system, so they have the zaptel style names). My files are below - the thing to note is the span 1,1,0, the second 1 tells you that the span is a timing source, externally clocked. Depending on the mode that your Ericsson is in, you may need to change signalling=pri_cpe to signalling=pri_net /etc/asterisk/zapata.conf: ; Configuration file [channels] ; ; Default language ; language=en context=ntl switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes group=1 callgroup=1 pickupgroup=1 ;echocancel=256 ;channel = 1-6 channel = 1-15,17-31 and /etc/zaptel.conf : span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone = uk On 16 Mar 2009, at 18:11, Oguzhan Kayhan wrote: Hello, I am trying to install my E1 card to make a conection with an Ericsson MD-110 PBX. I installed dahdi drivers as: dahdi_hardware pci::04:08.0 wcte12xp-d161:8000 Wildcard TE121 ran dahdi_genconf and it created all my e1 ports. On the other side i also configured the pbx to communicate with TE121. On ericsson side, i have no error messages. On asterisk side, no error messages. But when i try to create a dahdi trunk, and dial it from asterisk , no call can be made. and also, when i try to call from ericsson side, i get line busy message as soon as i dial the number. Is there any guide that can help me in installing that card? PS: Whatever i made in SPAN config, everytime the only thing i see was Internal clock on dahdi_tool . How can i make my e1 card master (or slave whatever) instead of internal clock?? and other thing i wonder, if i create a span like span=1,0,0,ccs,hdb3 is it zap/g1 in zaptel(dahdi) conf menu in asteriskgui???(or freepbx) ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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] Help Inbound number
On Mon, 16 Mar 2009, Bayardo Sanchez wrote: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed I run 1.2 so the command syntax may be different... 1) Enter sip show users to list your users. 2) Note the Def.Context for the user you are receiving this call from. 3) Enter show dialplan context where context is the context from the preceding step. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Busy on SIP
On Mon, 16 Mar 2009, Olivier wrote: 2009/3/16 Gordon Henderson gordon+aster...@drogon.netgordon%2baster...@drogon.net On Mon, 16 Mar 2009, Marco Sambo wrote: Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. Disable call-waiting inside the phone. Doesn't call-limit=1 force the same behaviour ? It appears to limmit the number of outgoing calls from that phone and independantly the number of inoming calls. So a phone can make an outgoing call, and still take an incoming call, and vice-versa, with call-limit=1 I also found early versions of this buggy in that it didn't seem to properly decrement the counter on hang-up, so is call-limit was set to 3, then that phone could only take 3 calls, one after the other, before it would be premenantly busyd, but this was a long time back, and it might have been something I was foing, but since then I always turned call-waiting off on the phones when users didn't want multiple call features. 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
Re: [asterisk-users] Help Inbound number
On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.com wrote: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. On Mon, 16 Mar 2009, Bayardo Sanchez wrote: [default] exten = 1246463,1,Answer(SIP/8003) 246463 != 1246463 Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help Inbound number
1246463 is not the same as 246463. Note the missing 1 If you want to match what is being dialed then your extensions.conf should look like this: [default] exten = 246463,1,Answer(SIP/8003) Bayardo Sanchez wrote: in my extension.conf i set : [default] exten = 1246463,1,Answer(SIP/8003) On Mon, Mar 16, 2009 at 12:06 PM, Pascal Bruno tipas...@gmail.com mailto:tipas...@gmail.com wrote: Do you have an extension set for 246463 in your extensions.conf? On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com wrote: i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com mailto:bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com mailto:bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] ATA react to phone but unresponsive to fax modem
Hi, I'm rather new to this domain so I may be doing stupid things without being concious of that. I've got a Patton MATA I'm trying to setup as T.38 fax adapter. Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting No signal. Line is busy or disconnect from Windows XP fax application. Whatching SIP trafic from this Patton MATA, I can see no single SIP is leaving the box so I'm certain issue relates to analog line settings but I'm mostly lost with things like Ring Polarity, Ring settings and so on. I tried to mimic settings from an SPA3102 with which I can either fax from fax machine or fax application but I'm unsuccessful at the moment. 1. Can you explain what is going on ? 2 What would you say reading this : Ring waveform: trapezoid Ring frequency: 20 Ring voltage: 85 FXS input gain: -6 FXS output gain: -6 (I copied those values from SPA3102 into MATA) Best 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] Help Inbound number
are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Help Inbound number
Bayardo Sanchez wrote: in my extension.conf i set : [default] exten = 1246463,1,Answer(SIP/8003) This should be: exten = 246463,1,Dial(SIP/8003) 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] Ast/Hyla/IAX Scalability?
If you using cisco why don't you use fax on/off ramp it works quite well. Then you can do with the fax file whatever you want. From other point of view I did connected 1.6.0.6 with spandsp-0.0.5 to PRI and receivefax seems to be working ok. The connect speed is low somewhere between 2400-9600 but it seems to be working. Actually I was able to receive international fax. Of course with some failures :-) If you want to use T38 in asterisk over ip with ata I didn't have too much luck with it. May be it would worked better on LAN. I switched to cisco or other hardware and it worked ok. Vladimir -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Underwood Sent: Friday, March 13, 2009 10:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ast/Hyla/IAX Scalability? David Backeberg wrote: On Fri, Mar 13, 2009 at 2:30 PM, Marshall Henderson marshall...@gmail.com wrote: On Fri, Mar 13, 2009 at 11:07 AM, David Backeberg dbackeb...@gmail.com wrote: Again, you'll find people arguing that their voip solution has as low of a failure rate as a hardware solution. I'm jealous. My voip fax solution does not yet have that low of a failure rate, but I'm hopefully getting closer to working out the last bugs. Do you have any specifics to share about the problems you're finding? Sure. I can't disagree with the poster who said that problems they've seen are really the other side's fault. But assigning blame doesn't make me any happier. I have fax receiving problems I can't reproduce. When I load test it, I don't have problems. When I send 'real' inbound faxes from outside the network, over the real phone system, I don't have problems. I'm in New Haven, CT. One sender that messes up the most is in Kansas City, KS. They are a legitimate client, really sending a fax. I get occasional fax receipts that say: 'The call dropped prematurely' There will sometimes be a cluster of these, followed by a successful receipt. When I load tested, and send from real fax machines out and back in on POTS, I get 100% success. I've successfully load-tested around 175 simultaneous inbound faxes. I slowed down the simulation to about 5 simultaneous faxes, and left that running over a long weekend, generating something like 30,000 faxes and something like 1GB of received fax files. Again, the success rate was 100%. A problem with my simulation was that I used sending faxes that speak the protocol correctly. Does anybody have some faxes that send garbage? Then I put it into production with a limited amount of real fax traffic for our clients. I'm talking fewer than 10 calls per day most days. But it seems like the reality of the speed of light over continental long-distance, combined with the reality of crappy fax machines that don't speak protocols correctly result in occasional failures. I've made some adjustments that I think anecdotally have solved the silly problems, but that one with the faxes dropping early is the one that (maybe) hasn't gone away. I'd like a success rate around 99%. I'm getting around 63% if you count individual failed calls that eventually result in a success. I can't tell if I'm having bad luck with this phase of my pilot or if my failure rate is going to remain constant as I add clients. I need more data points to get statistical significance. What I really need is a failing fax I can control, then tune parameters on my side, and see if the failure rate gets worse or better. Seriously considering breaking down and asking for the cooperation of the client in that endeavor. People who have been following my posts on this topic know that I'm using: PRI(s) - Cisco voip gateway hardware - T.38 / SIP / g711 - Asterisk-1.6 with ReceiveFax (depends on SpanDSP, but does NOT use IAX or IAXmodem) What I've been 'tuning' most recently have been arguments to the Cisco setup fax and SIP translation. I did try out IAXModem with Hylafax and 1.4 and had lots of problems that all went away when I switched to using the approach I use now. I never tried 1.6 with IAXModem and Hylafax, so I can't tell you how well they work together. Fully open-to-the-public FAX servers tend to get just get a lot of bad calls, many of them wrong numbers, or voice users. FAX servers for closed user groups tend to get few bad calls, unless the phone number gets included on some unfortunate list. This is one of the things which made early real world testing of spandsp and iaxmodem tough. We have to capture every failure, and analyse them by hand whether it was our fault or the far end's. Without knowing the nature of your system I have no clue what kind of failure rate might be expected. You can find a bit more about these issues and our results at http://www.soft-switch.org/spandsp-soft-fax-performance.html Your differing failure
Re: [asterisk-users] work around the 64 pickupgroups limit
At 22:22 3/13/2009, Matt Riddell wrote:
On 14/03/2009 10:29 a.m., Doug wrote:
At 16:10 3/10/2009, Matt Riddell wrote:
On 7/03/2009 4:58 a.m., Klaus Darilion wrote:
Hi!
What are the typical ways to work around the 64 groups limit?
What we actually do is store a pickup group with a caller id.
So the AsteriskDB has ${DB/pickup/${CALLERID(num)}} and we set
pickupmark to the same.
That way when someone dials 29 (what we use for pickup) it just checks
that group - no limitations on number of groups that way.
Hey Matt,
Would share some config file code with us?
Hi Matt,
This looks great! A few questions...
in the standard extension macro we add a line:
Is this in extensions.conf?
exten = s,n,Set(_PICKUPMARK=${DB(pickupgroup/${ARG1})})
Where ARG1 is the extension about to be called (i.e. 201)
When someone dials 29 to pickup:
exten = 29,1,Pickup(${DB(pickupgroup/${CALLERID(number)})}...@pickupmark)
Would this also be in extensions.conf?
So to make extension 201 in pickup group 1 just do:
asterisk -rx 'database put pickupgroup 201 1'
So this is a command line argument. Can this
be automated? Whenever we do a reload, can
this be stored?
--
Kind Regards,
Matt Riddell
Director
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[asterisk-users] Asterisk is not designed for University with large user base?
Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\012' ___ -- 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 is not designed for University with largeuser base?
Sounds like a personal preference to me. Here is the Wiki for SipX. http://en.wikipedia.org/wiki/SipX Reading this, it's just another flavor of the same medicine. Both are open-source with Commercial support available. In the 3 month's I've been reading this forum, there have been discussions of installations that are at least equivalent to a 10K user university. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vincent Li Sent: Monday, March 16, 2009 4:34 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk is not designed for University with largeuser base? Hello, I just had a meeting about a pilot project going on in our University, The project manager has done some research in the past year and concluded that Asterisk can not scale well to large user base like 10,000 users, thus Asterisk is not fit for large University environment. The project manager instead choosed sipX and said it scales well for large user base. I had an Asterisk running in my office for small user base, I don't have experience with large scale Asterisk implementation. I know little about sipX. Does anyone in the community has any input about this? Vincent Li System Administrator BRC,UBC perl -e'print\131e\164\040\101n\157t\150e\162\040\114i\156u\170\040\107e\145k\01 2' ___ -- 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] Help Inbound number
nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Help Inbound number
Just to read this right you are trying to take an inbound call from 888xxx and transfer it to your sip extension 8003? If so, Are you able to make internal calls to 8003? Can you transfer other calls to 8003 (exten = s,1,Dial(SIP/8003) ) ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo Sanchez Sent: Monday, March 16, 2009 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help Inbound number nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] T.38 - Which endpoint shall reINVITE ? caller or callee ?
Hi, I've been playing with T.38. I observed that mostly but not always, it's the calling endpoint that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the standardized or most common, way to start a T.38 session ? Shall it come from callee or from caller ? 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] Help Inbound number
The inbound was working well suddenly stopped working I want all calls made to the number should answer the extension 8003 On Mon, Mar 16, 2009 at 3:49 PM, Danny Nicholas da...@debsinc.com wrote: Just to read this right – you are trying to take an inbound call from 888xxx and transfer it to your sip extension 8003? If so, Are you able to make internal calls to 8003? Can you transfer other calls to 8003 (exten = s,1,Dial(SIP/8003) ) ? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Monday, March 16, 2009 4:38 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Help Inbound number nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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] Good phone near $125
I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
Mmm, $100-$125 What? USD? CAD? AUD? If you're willing to a little bit more, I'll strongly recommend Polycom IP 430. We're using them and they are absolutely painless (well, except the initial package of 100 of those which were heavy and caused some back pain ;p) Singer David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- 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] Help Inbound number
Your sip.conf should look like this sip.conf [procall] type=peer username=XX secret=XX context=default and extensions.conf [default] exten = 246463,1,Dial(SIP/8003) you must also have a sip user for 8003 in your sip.conf like [8003] type=friend username=XX secret=XX context=outgoing And dont forget to do a sip reload and dialplan reload On Mon, Mar 16, 2009 at 6:23 PM, Bayardo Sanchez bayardo.sanc...@gmail.comwrote: The inbound was working well suddenly stopped working I want all calls made to the number should answer the extension 8003 On Mon, Mar 16, 2009 at 3:49 PM, Danny Nicholas da...@debsinc.com wrote: Just to read this right – you are trying to take an inbound call from 888xxx and transfer it to your sip extension 8003? If so, Are you able to make internal calls to 8003? Can you transfer other calls to 8003 (exten = s,1,Dial(SIP/8003) ) ? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez *Sent:* Monday, March 16, 2009 4:38 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Help Inbound number nothing the problem persitem On Mon, Mar 16, 2009 at 12:42 PM, Geraint Lee gera...@gmail.com wrote: are you sure calls from this provider are going to context 'default' ? sip.conf [procall] type=peer username=XX secret=XX context=default 2009/3/16 Bayardo Sanchez bayardo.sanc...@gmail.com i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '246463' rejected because extension not found. but the extensin existed -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- 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 -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received
Re: [asterisk-users] Good phone near $125
We have used SNOM 360s, @ about $200, but just tried some Grandstream GXP2000. I like the 360s but the Grandstream is only $79.00, has four lines, good speaker phone, and will use a $10 cell headset. YMMV. But it works, and the price is right. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Monday, March 16, 2009 5:25 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Good phone near $125 I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 ___ -- 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] T1 problem (call using a .call file)
I have a weird problem with call using my T1 card. I can make calls fine using my analog and IP phones, but when I try to initiate a call using a .call file, I get the following error -- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- PROGRESS with cause code 127 received it happens on certain numbers I dial, but if I dial that same number with an ip or analog phone that use the T1 channel, the call is going through normally. Anybody knows why? My call file looks like this: Channel: DAHDI/g1/1XX Callerid: XX MaxRetries: 1 RetryTime: 5 WaitTime: 60 Context: test Extension: s Priority: 1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Uptime for documentation only
I know there has been better uptime than this reported, but I figured I'll share it anyhow: @pbx:~# uptime 18:39:07 up 621 days, 9:40, 2 users, load average: 0.00, 0.00, 0.00 pbx:~# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1200MHz stepping: 9 cpu MHz : 1197.305 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm pni est tm2 rng rng_en ace ace_en bogomips: 2398.50 pbx:~# cat /proc/meminfo MemTotal: 905744 kB MemFree:447660 kB Buffers: 3 kB Cached: 201616 kB SwapCached: 0 kB Active: 259584 kB Inactive: 101876 kB HighTotal: 0 kB HighFree:0 kB LowTotal: 905744 kB LowFree:447660 kB SwapTotal: 0 kB SwapFree:0 kB Dirty: 128 kB Writeback: 0 kB Mapped: 137844 kB Slab:94404 kB CommitLimit:452872 kB Committed_AS: 153068 kB PageTables:536 kB VmallocTotal: 122840 kB VmallocUsed: 320 kB VmallocChunk: 122176 kB In addition it has a 4GB Flash IDE HDD and a Digium Single Span T1 Card connected to an Adit 600. Running around 15 extensions a mix of Polycom 501s 3 Linksys SPA-94x and one Aastra 480i (I hate that last one). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.24 Now Available!
The Asterisk Development Team is proud to announce release of Asterisk 1.4.24, and is available for immediate download at http://downloads.digium.com/ In addition to other bug fixes, this release candidate fixes several crash issues, and resolved some remaining issues related to call pickup and call parking that were discovered after the release of Asterisk 1.4.23. In addition, issues related to chan_iax2, and regressions introduced to the 'h' extension have been resolved. This release marks the first inclusion of the release summary files which will be included in all future releases. The purpose is to give a clearer overview of the changes that have taken place between the current and previous release, which issues have been closed, and which community members were involved with issue submission, code commits, and issue testing. Additionally, a diffstat at the end of the file shows at a brief glance the number of changes made to files between the previous and current releases. For a summary of the changes in this release, please see the release summary: http://svn.digium.com/view/asterisk/tags/1.4.24/asterisk-1.4.24-summary.html?view=co For a full list of changes in this release, please see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.4.24/ChangeLog?view=co The following list of bugs were resolved with the participation of the community, and this release would not have been possible without your help! * Paging application crashes asterisk - Closes issue #14308. Submitted by bluefox. Tested by kc0bvu. Patched by seanbright. * Crash in VoiceMailMain if hangup occurs before a valid mailbox number is entered (IMAP only) - Closes issue #14473. Submitted by, and patch provided by dwpaul. * Incoming Gtalk calls fail - Closes issue #13984. Submitted by, tested, and patched by jcovert. * Realtime peers are never qualified after 'sip reload' - Closes issue #14196. Submitted by, tested, and patched by pdf. * SIP Attended Transfer fails - Closes issue 14611. Submitted by, tested, and patched by klaus3000. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good phone near $125
Polycom On Mon, Mar 16, 2009 at 6:24 PM, David Ruggles da...@safedatausa.com wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.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 ___ -- 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] Best way to get 60+ analogue extensions.
Channel Banks would be the way I would do it. On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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] Good phone near $125
David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200da...@safedatausa.com I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. N. ___ -- 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] Best way to get 60+ analogue extensions.
I'll second that. On Mon, 16 Mar 2009 18:48:10 -0400, C F shma...@gmail.com wrote: Channel Banks would be the way I would do it. On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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] Best way to get 60+ analogue extensions.
I don't know how good Asterisk's GR.303 support, but you could use DLCs as well. However, that's a lot of complexity and (seemingly) immature functionality liability to achieve the same end you'd get with a channel bank. The only benefit is that DLCs are specifically for oversubscription, whereas on PRIs you'd be doing one timeslot per one POTS line on the trunk side. On Mon, 16 Mar 2009 18:48:10 -0400, C F shma...@gmail.com wrote: Channel Banks would be the way I would do it. On Sun, Mar 15, 2009 at 3:12 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- 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 is not designed for University with largeuser base?
Danny Nicholas wrote: Sounds like a personal preference to me. Here is the Wiki for SipX. http://en.wikipedia.org/wiki/SipX Reading this, it's just another flavor of the same medicine. Both are open-source with Commercial support available. I'd contend that the business model says very little about implementation, reliability, scalability. ___ -- 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] Good phone near $125
On Mar 16, 2009, at 3:53 PM, SIP wrote: David Ruggles wrote: I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 da...@safedatausa.com I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. They do, we have a bunch of 300's (and 320's) deployed as PoE. Daniel N. ___ -- 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] ATA react to phone but unresponsive to fax modem
2009/3/16 Olivier oza-4...@myamail.com Hi, I'm rather new to this domain so I may be doing stupid things without being concious of that. I've got a Patton MATA I'm trying to setup as T.38 fax adapter. Whenever I connect a fax machine (Dell MFP1815dn) or a phone to it, I can successfully send a fax or talk to the other end. Whenever I connect a fax modem (Dell Inspiron 6400 laptop), I keep getting No signal. Line is busy or disconnect from Windows XP fax application. Whatching SIP trafic from this Patton MATA, I can see no single SIP is leaving the box so I'm certain issue relates to analog line settings but I'm mostly lost with things like Ring Polarity, Ring settings and so on. I tried to mimic settings from an SPA3102 with which I can either fax from fax machine or fax application but I'm unsuccessful at the moment. 1. Can you explain what is going on ? 2 What would you say reading this : Ring waveform: trapezoid Ring frequency: 20 Ring voltage: 85 FXS input gain: -6 FXS output gain: -6 (I copied those values from SPA3102 into MATA) Best regards Changing FXS input gain and FXS output gain from -6 to -12 improved things as I could fax out in T.38 with both ATAs and fax endpoints ! But for incoming faxes, modem connected to M-ATA remains silent and idle whenever the M-ATA receives a fax call : I can see incoming SIP signal arriving into the ATA but it seems no analog signal is going out from it. (using SPA3102, faxes are correctly received). How is called the signal an ATA uses when it wants to wake an analog phone or a fax machine up ? Is it correct to think the same electrical signal is sent whatever the analog endpoint is ? What could explain a phone is ringing at one and a fax modem remains idle ? 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] Good phone near $125
SIP wrote: I believe SNOM 300s do PoE (might have to check that, though) and are around $100. We've little experience with them, but we use an office full of Snom 320s, and we're nothing but pleased with them. Good speaker, good handset, lots of excellent options. And reasonably priced. N. The first generation of Snom 300's did _not_ support POE - but later models did. PaulH ___ -- 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] Multi-tenant with receptionist features for managed service
Dear all, I'm currently researching options for a MT asterisk gui/system for a small business centre that will have 12 units in it. Each unit will be configured for one extension. The system there will have a max of 12 concurrent calls to PSTN provided via an ADSL/SDSL link to our VoIP provider in the UK, using g.711, maybe g.729 dependant on networking costs. Fallback will be to 4 analogue lines should this go down. What is key is billing information and the ability for a receptionist to see all active calls and do transfers etc. Much like the Flash Operator Panel. Desktop Software may also be needed for this purpose or can be done via a traditional bank of lines on an IP phone accessory module. If anyone has any ideas on the best way to put this together, I'm all ears ;-) I was going to use an OpenVOX card and Dell T100 box, with 12 Aastra 53i phones. There's a £4k budget for this (still waiting for more into)which will include the networking connection and equipment. If I can afford it I normally go Sangoma with Echo cancellation, but as it's a fallback service, so I'm not bothered. I think I've covered everything. There will be many more business centres to come as this first project will be the blueprint one. The end goal is to also move this to a data centre and not have it on site with the pstn fallback options, but use redundant links to our DC. Like a mini-ITSP for our area. I haven't figured the receptionist part for that bit yet though ;-) Thanks, Gavin. P.S. I have thought about pbxinaflash and a2billing, but I'm not sure if it would not be clunky for a novice to handle (receptionist). I may go down that route and hire the FreePBX team to fill in the mixing pieces of Multi-tenant if they are interested. ___ -- 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] Problem with Verizon Wireless
Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- 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] Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem)
Hi,
Is the following behaviour a bug or a feature ?
Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces :
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least
one of app or extension (or keyword message/pdu) must be specified, along
with tech and dest in file /var/spool/asterisk/outgoing/astup.call
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:457 scan_service: Invalid file
contents in /var/spool/asterisk/outgoing/astup.call, deleting
[Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:505 scan_thread: Failed to
scan service '/var/spool/asterisk/outgoing/astup.call'
Reverting Bristuff's patch on lines 266 in asterisk/pbx/pbx_spool.c such as
correct this and call file can be played :
/* Original code
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten)) ||
(ast_strlen_zero(o-message) ast_strlen_zero(o-pdu))) {
*/
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten))) {
Is this a feature or a bug (ie do you have to either add a pdu or a message
(or both) in call files when using Bristuff) ?
Regards
___
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Re: [asterisk-users] Problem with Verizon Wireless
Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. 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] Good phone near $125
I was looking at the aastra 9133i, however I was informed that this phone is no longer supported. What are good phones around the $100 - $125 price point? (Need POE) I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet, support PoE and works with 2.5mm headset. $110 at voipsupply ___ -- 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] Ast/Hyla/IAX Scalability?
VB wrote: If you using cisco why don't you use fax on/off ramp it works quite well. Then you can do with the fax file whatever you want. From other point of view I did connected 1.6.0.6 with spandsp-0.0.5 to PRI and receivefax seems to be working ok. The connect speed is low somewhere between 2400-9600 but it seems to be working. Why do people love to just make stuff up when they post? The range is 4800 to 14400, and that's the full range of most FAX machines. Some go up to 33600, but there are lots of problems, and the required modem is encumbered, so a free solution isn't possible. Actually I was able to receive international fax. Of course with some failures :-) Why of course? If you want to use T38 in asterisk over ip with ata I didn't have too much luck with it. May be it would worked better on LAN. I switched to cisco or other hardware and it worked ok. A *lot* of ATAs have very buggy T.38, if they have it at all. Gateways aren't much better. You'll find if you try a few Cisco firmware revisions that T.38 peformance varies considerably. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
Good luck having Verizon change that. In the meantime why don't you try implementing a call screen feature so that the call is not considered answered until a key is pressed by the one answering? That way the caller will still hear ringing until the one answering presses that key. On Mon, Mar 16, 2009 at 7:27 PM, drew einhorn drew.einh...@gmail.com wrote: Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- 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] Problem with Verizon Wireless
On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. 4 channels? Could you count them for me please? I'm just getting started and working my way up from the simplest configurations. I may not have the jargon right right. I was expecting that I could eventually configure things so that I could hand off the calls so that once the Asterisk box got a connection between the DID provider originating the call and whatever/whoever is terminating the call (SIP device, or SIP service provider) the Asterisk box could then drop out of the connection and let the originator talk directly to the terminator. Is this an unrealistic assumption. Ah, I see one disconnect. I think you are assuming T1 or better connections to the PSTN where you are originating and terminating the calls yourself and I'm using SIP service providers to do all the origination and termination. I'm connecting a bunch of home offices scattered around the country and do not have enough lines in any city to justify originating or terminating my own PSTN calls. Maybe just one PSTN line per DSL connection to avoid paying a sip provider to terminate some local calls, and supporting some backup functionality, if the Asterix box has crashed, but it will be a while before things get that complicated. ___ From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- 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 Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. 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 -- Drew Einhorn ___ -- 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] system sizing
I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
On Mon, Mar 16, 2009 at 6:47 PM, C F shma...@gmail.com wrote: Good luck having Verizon change that. In the meantime why don't you try implementing a call screen feature so that the call is not considered answered until a key is pressed by the one answering? That way the caller will still hear ringing until the one answering presses that key. Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. Or, am I misunderstanding you suggestion. -- Drew Einhorn ___ -- 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] system sizing
At 19:57 3/16/2009, Eric Fort wrote: I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric Most new hardware is severe overkill for Asterisk--it's just not that resource intensive. This would work: ~~ Codegen case: http://www.newegg.com/Product/Product.aspx?Item=N82E16811182566 Thermaltake power supply: http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=BA23480 Motherboard: Gigabyte GA-M61PME-S2 http://www.google.com/products/catalog?hl=enq=M61PME-S2cid=8677888669212799391scoring=mrd#ps-sellers AMD CPU- Athlon X2 5050E AM2 2.6GHZ 1MB 65NM 45W 2000MHZ Pib: http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA25671 2GB Crucial Memory: http://www.mwave.com/mwave/skusearch.hmx?scriteria=BA24642 Seagate Harddrive 500 GB: http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=AA72270 Pioneer CD/DVD: http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=AA73666 Floppy Drive: http://www.mwave.com/mwave/viewspec_v2.asp?scriteria=AA00693 Extra case fans: http://store4pc.stores.yahoo.net/80ulquietcas1.html ~~ Right now it's important to support AMD. If they go under, Intel will just slack off. If you want to keep it *very* simple: http://PBXinaFlash.net/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
I have a possible suggestion -- don't consider the call answered unless someone types a 1 or something -- makes the dial plan more complex, but it should work pretty well. on Monday 03/16/2009 drew einhorn(drew.einh...@gmail.com) wrote On Mon, Mar 16, 2009 at 6:47 PM, C F shma...@gmail.com wrote: Good luck having Verizon change that. In the meantime why don't you try implementing a call screen feature so that the call is not considered answered until a key is pressed by the one answering? That way the caller will still hear ringing until the one answering presses that key. Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. Or, am I misunderstanding you suggestion. -- Drew Einhorn ___ -- 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 -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] system sizing
I may not need a 1 to 1 ratio of phones to sip channels Mostly I'm trying to be a bit conservative on my estimates to leave plenty of room for expansion and growth. I'm hoping that I won't need to do much transcoding (but I shouldn't as long as the phones and the ITSP use the same codec). a few of the inbound channels would be for access to voicemail and calls on them would likely never reach a phone. Eric On 3/16/09, clemen...@gmail.com clemen...@gmail.com wrote: 3GHz P4 with a gig will do that just admirably... Disk space required for voice mail, would vary on your config. Do you really need a 1:1 ratio of SIP channels to phones (assuming you mean inbound from a SIP provider) ~Max --Original Message-- From: Eric Fort Sender: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: eric.f...@gmail.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] system sizing Sent: Mar 16, 2009 17:57 I'm looking to install a basic asterisk system for my church with: 8 inbound sip channels 8 sip handsets basic voicemail room to grow (maybe doubling each of the above) What would be a recomended system as to needed processor and memory? Thanks, Eric ___ -- 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 Sent from my BlackBerry device on the Rogers Wireless Network ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
drew einhorn wrote: snip Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. On T-mobile, a subscriber can choose to forward to another number in the PSTN if the phone cannot be found, they call unavailable, and even a different number if the cell doesn't answer in a number of rings ( AFAIK not adjustable by the subscriber ) This can all be done a the mobile phone, so no customer service help required. Unknown if VeriZon can do the same. Questionable if customer service can answer either, if they are as good as most! John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
Which if you follow my solution will still ring to the other phones/devices. http://www.voip-info.org/wiki-Asterisk+cmd+Dial Use the M option to accomplish this (I'm 1.2 here) if you use 1.4/1.6 then there might be an easier solution, not sure. On Mon, Mar 16, 2009 at 8:58 PM, drew einhorn drew.einh...@gmail.com wrote: On Mon, Mar 16, 2009 at 6:47 PM, C F shma...@gmail.com wrote: Good luck having Verizon change that. In the meantime why don't you try implementing a call screen feature so that the call is not considered answered until a key is pressed by the one answering? That way the caller will still hear ringing until the one answering presses that key. Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. Or, am I misunderstanding you suggestion. -- Drew Einhorn ___ -- 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] Problem with Verizon Wireless
John Novack wrote: drew einhorn wrote: snip Maybe I don't understand this suggestion. I think your suggestion applys to my sip phones/atas, but they are not the problem. The problem is that when Verizon's network notices the the cell phone is currently not on their network, they pick up the call and answer with a voice error message (sometimes after only one ring), before anybody has a chance to answer on a sip device. On T-mobile, a subscriber can choose to forward to another number in the PSTN if the phone cannot be found, they call unavailable, and even a different number if the cell doesn't answer in a number of rings ( AFAIK not adjustable by the subscriber ) This can all be done a the mobile phone, so no customer service help required. Unknown if VeriZon can do the same. Questionable if customer service can answer either, if they are as good as most! John Novack I just spoke with a VeriZon wireless tech who maintains cell sites. VeriZon wireless network can have calls forwarded on no answer or immediate to other than their voice mail, but there seems to be no way to escape the network recording if the phone can't be found. You may want to try forwarding to another PSTN number either on now answer or immediate and see if he is mistaken John Novack -- Dog is my co-pilot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Verizon Wireless
Is the feature you are implementing Single Number Reach? They dial a number and you call another number (Verizon Cell Phone) trying to connect them to the user? But the problem is Verizon answers with the silly out of reach message? I've never seen where the PSTN carrier lets you re-direct the call to the cell phone without your Single Number Reach PBX holding/hairpinning the call. I'm more old school PBX than SIP expert and suspect this can be done in the SIP cloud. I suspect services like Vonage Ring Lists don't hairpin calls! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Monday, March 16, 2009 8:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with Verizon Wireless On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. 4 channels? Could you count them for me please? I'm just getting started and working my way up from the simplest configurations. I may not have the jargon right right. I was expecting that I could eventually configure things so that I could hand off the calls so that once the Asterisk box got a connection between the DID provider originating the call and whatever/whoever is terminating the call (SIP device, or SIP service provider) the Asterisk box could then drop out of the connection and let the originator talk directly to the terminator. Is this an unrealistic assumption. Ah, I see one disconnect. I think you are assuming T1 or better connections to the PSTN where you are originating and terminating the calls yourself and I'm using SIP service providers to do all the origination and termination. I'm connecting a bunch of home offices scattered around the country and do not have enough lines in any city to justify originating or terminating my own PSTN calls. Maybe just one PSTN line per DSL connection to avoid paying a sip provider to terminate some local calls, and supporting some backup functionality, if the Asterix box has crashed, but it will be a while before things get that complicated. ___ From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower. Verizon searches their network and if they cannot find the cell phone, they pick up the call and generate a voice error message. Or if the cell lines are busy they generate busy signal. I need to know the right incantation to use with Verizon to get them to just let the cell lines ring until either some picks up a voip line, or the voip voicemail picks up the call. -- Drew Einhorn ___ -- 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 Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] Plastic Water Bottles
The plastics industry says polycarbonate bottles are safe. http://www.bisphenol-a.org/about/faq.html#g I'm sure Maggie and here friends would say ALL plastic bottles are very dangerous. This lady seems to be at a reasonable middle ground. http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water_bottles Polycarbonate plastics the kind of bottle you bought contains BPA. In 2006 Europe banned all products made for children under age 3 containing BPA, and as of Dec. 2006 the city of San Franscisco followed suit. In March 2007 a billion-dollar class action suit was commenced against Gerber, Playtex, Evenflo, Avent, and Dr. Brown's in Los Angeles superior court for harm done to babies caused by drinking out of baby bottles and sippy cups containing BPA. So, to be certain that your baby is not exposed, use glass bottles She suggests that if you really want a plastic bottle, get ones made from a different kind of plastic. http://www.nalgene-outdoor.com/store/SearchResult.aspx?CategoryID=10 But she really recommeds storing water in glass, brass, or ceramic bottles. She does not discuss aluminum. Here's another author on aluminum. The problem with aluminum is. What is the inside of the water bottle lined with? Sigg water bottle from Switzerland are the highly recommended, but expensive. Here's a Sigg clone for $12.00 http://www.everythingyoga.com/colored-water-bottle-18oz.htm Here's a really ugly link to google ads for non bpa water bottles. http://googleads.g.doubleclick.net/pagead/ads?client=ca-pub-0918706375590523dt=1237256368148lmt=1237255974format=fp_al_lpoutput=htmlcorrelator=1237256368148channel=1097421959url=http%3A%2F%2Ftrusted.md%2Fblog%2Fvreni_gurd%2F2007%2F03%2F29%2Fplastic_water_bottlesad_type=text_imageea=0ref=http%3A%2F%2Fwww.google.com%2Fsearch%3Fq%3Dpolycarbonate%2Bleach%2BBPA%26ie%3Dutf-8%26oe%3Dutf-8%26aq%3Dt%26rls%3Dcom.ubuntu%3Aen-US%3Aunofficial%26client%3Dfirefox-afrm=0ga_vid=1194625357321842000.1237255497ga_sid=1237255497ga_hid=600499628ga_fc=trueflash=10.0.22u_h=1200u_w=1600u_ah=1200u_aw=1600u_cd=24u_tz=-360u_his=13u_java=trueu_nplug=15u_nmime=155dtd=16kw_type=radlinkprev_fmts=728x15_0ads_alrt=ChBJvwi1AAQc7AqDJwTq9m-vEhZCUEEgRnJlZSBXYXRlciBCb3R0bGVzGgjBPl5rQmDuSygBUhMI8aH_lPOomQIVBE6DCh2MHDrkhl=enkw0=BPA+Free+Water+Bottleskw1=Plastic+Water+Bottles+7kw2=Bisphenol+a+Bottleskw3=Plastic+Sippy+Cupsokw=BPA+Free+Water+Bottles -- Drew Einhorn ___ -- 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] Plastic Water Bottles
On Monday 16 March 2009 21:49:53 drew einhorn wrote: snip What does this have to do with Asterisk? -- 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
Re: [asterisk-users] Problem with Verizon Wireless
On Mon, Mar 16, 2009 at 8:45 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: Is the feature you are implementing Single Number Reach? They dial a number and you call another number (Verizon Cell Phone) trying to connect them to the user? But the problem is Verizon answers with the silly out of reach message? I've never seen where the PSTN carrier lets you re-direct the call to the cell phone without your Single Number Reach PBX holding/hairpinning the call. I'm more old school PBX than SIP expert and suspect this can be done in the SIP cloud. I suspect services like Vonage Ring Lists don't hairpin calls! I'm just getting started in this are and learning the jargon (had to google, Single Number Reach, and hairpinning). Yes, I am trying to implement Single Number Reach. I'm really not ready to deal with hairpinning. I think that means the call comes into my system from the originator, the makes a sharp U-turn sort of like a hairpin shape an goes out to wherever the call is terminated. I believe, but I could easily be wrong, that with sip I can let go of the hairpin and let the sip originator talk directly to the sip terminator and get the asterisk box out of the picture once the call is properly connected. But I'm not yet ready to work on that part. My problem is that the Verizon network grabs the call and effectively says: it's mine, and I can't handle it. When Verizon should just ignore the calls they can handle, and let those who can handle the call, handle it. I've got to go take a closer look at some earlier comments that I did not quite understand on first reading. I may have to make the process of answering a call more complicated for the users. They have to answer the phone and press a key on the keypad to prove they are a human and not a stupid Verizon robot that had no business answering the phone. Arghhh!!! That's really ugly from a human interface stand point. And I've got to figure out how to implement it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of drew einhorn Sent: Monday, March 16, 2009 8:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem with Verizon Wireless On Mon, Mar 16, 2009 at 5:43 PM, Jason Aarons (US) jason.aar...@us.didata.com wrote: Nextel does that, pickups up after x rings and says 'The Nextel subscriber you are trying to reach is unavailable, please try your call again later. I'm not sure what Verizon or Nextel called this feature or what advantage is it for the carrier to play it versus just letting it ring forever... In general I've had similar issues, customers want voicemail and single number reach delivers the call to the device that answers, be it a home answering machine, cell phone voicemail, etc. I haven't had a customer keep single number reach as one call in can burn 4 or more channels out to each device. Doesn't scale real well. 4 channels? Could you count them for me please? I'm just getting started and working my way up from the simplest configurations. I may not have the jargon right right. I was expecting that I could eventually configure things so that I could hand off the calls so that once the Asterisk box got a connection between the DID provider originating the call and whatever/whoever is terminating the call (SIP device, or SIP service provider) the Asterisk box could then drop out of the connection and let the originator talk directly to the terminator. Is this an unrealistic assumption. Ah, I see one disconnect. I think you are assuming T1 or better connections to the PSTN where you are originating and terminating the calls yourself and I'm using SIP service providers to do all the origination and termination. I'm connecting a bunch of home offices scattered around the country and do not have enough lines in any city to justify originating or terminating my own PSTN calls. Maybe just one PSTN line per DSL connection to avoid paying a sip provider to terminate some local calls, and supporting some backup functionality, if the Asterix box has crashed, but it will be a while before things get that complicated. ___ From: asterisk-users-boun...@lists.digium.com on behalf of drew einhorn Sent: Mon 3/16/2009 7:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem with Verizon Wireless Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a remote location where it is too far away from a cell tower.
Re: [asterisk-users] Plastic Water Bottles
Tilghman Lesher wrote: On Monday 16 March 2009 21:49:53 drew einhorn wrote: snip What does this have to do with Asterisk? I was thinking plastic bottles are just todays version of cups on a string. ___ -- 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] Plastic Water Bottles
Sorry, It has absolutely nothing to do with this list. It was intended for my wife and was accidentally sent to the wrong address. I really hope I have not offended folks that I really want to answer the on topic questions I am asking on this list. I'm very, very sorry. On Mon, Mar 16, 2009 at 9:20 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Monday 16 March 2009 21:49:53 drew einhorn wrote: snip What does this have to do with Asterisk? -- 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 -- Drew Einhorn ___ -- 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] Plastic Water Bottles
Dear Sir, I am intrigued by your ideas and would like to subscribe to your quarterly newsletter as well as your annual seminar and leadership conference. -- Sent from mobile device On Mar 16, 2009, at 10:49 PM, drew einhorn drew.einh...@gmail.com wrote: The plastics industry says polycarbonate bottles are safe. http://www.bisphenol-a.org/about/faq.html#g I'm sure Maggie and here friends would say ALL plastic bottles are very dangerous. This lady seems to be at a reasonable middle ground. http://trusted.md/blog/vreni_gurd/2007/03/29/plastic_water_bottles Polycarbonate plastics the kind of bottle you bought contains BPA. In 2006 Europe banned all products made for children under age 3 containing BPA, and as of Dec. 2006 the city of San Franscisco followed suit. In March 2007 a billion-dollar class action suit was commenced against Gerber, Playtex, Evenflo, Avent, and Dr. Brown's in Los Angeles superior court for harm done to babies caused by drinking out of baby bottles and sippy cups containing BPA. So, to be certain that your baby is not exposed, use glass bottles She suggests that if you really want a plastic bottle, get ones made from a different kind of plastic. http://www.nalgene-outdoor.com/store/SearchResult.aspx?CategoryID=10 But she really recommeds storing water in glass, brass, or ceramic bottles. She does not discuss aluminum. Here's another author on aluminum. The problem with aluminum is. What is the inside of the water bottle lined with? Sigg water bottle from Switzerland are the highly recommended, but expensive. Here's a Sigg clone for $12.00 http://www.everythingyoga.com/colored-water-bottle-18oz.htm Here's a really ugly link to google ads for non bpa water bottles. http://googleads.g.doubleclick.net/pagead/ads?client=ca-pub-0918706375590523dt=1237256368148lmt=1237255974format=fp_al_lpoutput=htmlcorrelator=1237256368148channel=1097421959url=http%3A%2F%2Ftrusted.md%2Fblog%2Fvreni_gurd%2F2007%2F03%2F29%2Fplastic_water_bottlesad_type=text_imageea=0ref=http%3A%2F%2Fwww.google.com%2Fsearch%3Fq%3Dpolycarbonate%2Bleach%2BBPA%26ie%3Dutf-8%26oe%3Dutf-8%26aq%3Dt%26rls%3Dcom.ubuntu%3Aen-US%3Aunofficial%26client%3Dfirefox-afrm=0ga_vid=1194625357321842000.1237255497ga_sid=1237255497ga_hid=600499628ga_fc=trueflash=10.0.22u_h=1200u_w=1600u_ah=1200u_aw=1600u_cd=24u_tz=-360u_his=13u_java=trueu_nplug=15u_nmime=155dtd=16kw_type=radlinkprev_fmts=728x15_0ads_alrt=ChBJvwi1AAQc7AqDJwTq9m-vEhZCUEEgRnJlZSBXYXRlciBCb3R0bGVzGgjBPl5rQmDuSygBUhMI8aH_lPOomQIVBE6DCh2MHDrkhl=enkw0=BPA+Free+Water+Bottleskw1=Plastic+Water+Bottles+7kw2=Bisphenol+a+Bottleskw3=Plastic+Sippy+Cupsokw=BPA+Free+Water+Bottles -- Drew Einhorn ___ -- 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
