Re: [Asterisk-Users] hide caller id
try to put hidecallerid=no in your zapata.conf Pedro Vela wrote: Yes, my phone company has enabled the Caller ID hiden possibility, thats because with a Panasonic PBX works fine but with Asterisk not. Thanks for your aproach, what can I do now? Regards, Pedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] direct dial-in (DDI)
So, Can asterisk operate on DDI? There is DDI support for ISDN-CAPI cards that allow P2P mode. AVM B1 is supposed to do this, but it hung on my side. I've ordered cards with the HFC chipset and will try using zaphfc. As usual, www. voip-info.org should enlight you :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice and DTMF
It's official, Greg figured it out. And you know what, it all makes sense now: The scope for the dtmfmode setting is the section. Since the [broadvoice] section is needed for outgoing calls only, the [general] section -- the one containing the register directives would have to be where you define the dtmfmode for incoming connection. How about -- [general] dtmfmode=inband register = usera:[EMAIL PROTECTED] dtmfmode=rfc2833 register = userb:[EMAIL PROTECTED] Would that work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Sunday, June 13, 2004 5:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Greg, Per your suggestion, I added dtmfmode=inband to the general section of my sip.confthe other items you mentioned were already in sync with what I had. With that one change inbound DTMF to * IVR works! I will continue to play with it to flesh out it's reliability, but I was successfully able to navigate my IVR and log on to * VM. Thanks for the suggestion, I will followup with any interesting developments from my testing. Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill Sent: Sunday, June 13, 2004 4:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF On Sat, 12 Jun 2004, Jay Milk wrote: Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. I'm running CVS-HEAD-06/06/04. I've spent a couple hours tinkering and taking notes on the dtmf issue this morning. I tried various combinations of rfc2833 and inband in my dtmfmode= statements in sip.conf and with each combination tried dialling out (xten softphone - * - BV - cell phone voicemail) and calling in (cell phone - BV - * - IVR) to test DTMF functionality (or brokenness). During each call, I used show channel in the CLI to see how * really thought the channel was configured. I think I finally came up with a setup where DTMF works. I'm hoping maybe some of you who have been struggling with this issue also will give it a try and tell the rest of us if it works in your config. sip.conf: [general] ... dtmfmode=inband [broadvoice] type=peer ... dtmfmode=inband [xtenphone] type=peer ... dtmfmode=rfc2833 I don't have any allow/disallow statements for any codecs, although I'm thinking about bringing those into my puzzle soon.. Anyway, with sip.conf set up as I described above, I placed a call: xten - * - BV - cell phone and the DTMF was passed through so that I could interact with the voicemail system. In this call, * indicated: xten - * channel: codec=GSM, dtmfmode=rfc2833 * - BV channel: codec=ULAW, dtmfmode=inband When I tried to set the xtenphone to use dtmfmode=inband (in sip.conf), * filled my console with Unable to process inband DTMF on 2 frames and I couldn't capture any info on the channel setup through the CLI. Removing the dtmfmode=inband statement under [general] didn't affect results. So then I placed another call: cell phone - BV - * I set up extensions.conf so that the incoming call from BV would go into an IVR I built for controlling xmms. I was able to enter the extension numbers to control the system. I don't have voicemail set up on this * box, so I couldn't test a call to that app. In this call * indicated: BV - * channel: codec=ULAW, dtmfmode=inband Removing the dtmfmode=inband statement under [general] DID affect results! With this statement commented out, * indicates: BV - * channel: codec=ULAW, dtmfmode=rfc2833 In this setup, DTMF broke and I couldn't control my xmms. So.. Jay, Michael, others.. if you try this config, let us know what results you find! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP audio cut off even with Answer, Wait...
Ahh... We're doing the same thing! I just tested inbound DTMF with one of our T-Mo phones as well as with my spare Vonage line. On my T-Mo phone, the call is established instantly, and I get the friendly Comedian greeting. For testing purposes, all I have in the dial-plan for this number is: s,1,Answer s,2,Wait(1) s,3,VoiceMailMain() When using vonage on the other hand, I only get 'um to Comedian Mail'. I don't have a land-line to test here right now. Your question #4 is indeed a puzzle. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Sunday, June 13, 2004 11:08 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP audio cut off even with Answer, Wait... Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem. In googling for answers, I came across the recommendation to issue an Answer followed by a Wait of 2 or 3 seconds to let the SIP settle down before playing back audio. I do indeed have this in place; in fact I have been using that advice for quite a while now whether answering SIP or not I decided to play with the Wait times to see what would happen and this is what I found: Initially I was using Wait,3 - in this configuration whether by landline or cell phone, * would answer without any sort of ring indication at all - just silence from the time you finish dialling until asterisk starts to play some audio. I kept increasing Wait by 1 and testing with no difference either in the lack of ring indication, or the audio cutoff when using a cellphone until I reached Wait,7. At this point, there is still no ring indication, but the audio cutoff when calling via cellphone is fixed. Increasing to Wait,8 gives exactly 1 ring, then * answers and the audio is perfect calling from cell phone...each additional second of wait time above 8 gives additional ring indication, and perfect audio... 1) Should asterisk really take 7 seconds of wait for SIP to settle and not cut off audio? 2) Why would a call from a landline phone have no cutoff problems, even with a Wait,3? 3) I expected that there would either be no ringing indication at all, or that it would start immediately (or soon) after dialing. Why is there no ring indication unless you Answer then Wait,8? 4) If there is no ring indication prior to * answering, and I have no Ringing command configured in * anywhere in my extensions.conf, where is this ringing coming from? This has been a bit rambling, I apologize...any feedback greatly appreciated. Marty ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] collaboration with Panasonic PBX
Hi. I've searched the archives and found nothing regarding collaborating Asterisk with a Panasonic PBX (TD1232 to be exact) Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? On the hardware page for the X100P card is says it's great for handling incoming calls. It says nothing about making outgoing calls. Is it at all possible to use that card to make outgoing calls from Asterisk to the PSTN lines? Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hide caller id
Can't tell you how it works in Spain, but here in the US, we can dial
*67 before a phone number, and caller-id is hidden. I've had this
working with SIP before, by doing something like:
[macro-dialdomestic-private]
exten = s,1,Dial(SIP/*67${MACRO_EXTEN:[EMAIL PROTECTED],60)
exten = s,2,Dial(SIP/*67${MACRO_EXTEN:[EMAIL PROTECTED],60)
exten = s,3,Congestion
[dialNXX]
exten = _1NXXNXX,1,Macro(dialdomestic)
exten = _*671NXXNXX,1,Macro(dialdomestic-private)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedro Vela
Sent: Monday, June 14, 2004 12:47 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] hide caller id
Yes, my phone company has enabled the Caller ID hiden possibility, thats
because with a Panasonic PBX works fine but with Asterisk not. Thanks
for your aproach, what can I do now?
Regards,
Pedro
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Manuel Wenger
Enviado el: viernes, 11 de junio de 2004 10:36
Para: [EMAIL PROTECTED]
Asunto: R: [Asterisk-Users] hide caller id
Before starting to look at the problem in Asterisk, make sure that your
phone company has enabled the selective CLIR feature. Otherwise the
phone exchange will simply ignore your request to hide CLIP.
Regards
Manuel
-Messaggio originale-
Da: Pedro Vela [mailto:[EMAIL PROTECTED]
Inviato: venerdì, 11. giugno 2004 08:56
A: [EMAIL PROTECTED]
Oggetto: [Asterisk-Users] hide caller id
Hi,
We try ti hide the caller id at calls trought E1 in EuroISDN (Spain)
using restrictcid=yes and doesn´t work.
What can I do, thaks
Pedro
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Re: [Asterisk-Users] collaboration with Panasonic PBX
On the hardware page for the X100P card is says it's great for handling incoming calls. It says nothing about making outgoing calls. Is it at all possible to use that card to make outgoing calls from Asterisk to the PSTN lines? You can use it to make outgoing calls. Kannaiyan http://www.goods2world.com - Your VoIP Shop ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323
This module wont compile can anyone give me any assistance
RE: [Asterisk-Users] collaboration with Panasonic PBX
Thanks. -Original Message- From: Kannaiyan Natesan [mailto:[EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] collaboration with Panasonic PBX On the hardware page for the X100P card is says it's great for handling incoming calls. It says nothing about making outgoing calls. Is it at all possible to use that card to make outgoing calls from Asterisk to the PSTN lines? You can use it to make outgoing calls. Kannaiyan http://www.goods2world.com - Your VoIP Shop ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323
On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders [EMAIL PROTECTED] wrote: This module wont compile can anyone give me any assistance Sure, what error messages is it giving you Michael? -- -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] collaboration with Panasonic PBX
On Mon, 14 Jun 2004, Shoval Tomer wrote: Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? If you mean connecting the X100P to an analog extension line then that will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323
debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders [EMAIL PROTECTED] wrote: This module wont compile can anyone give me any assistance Sure, what error messages is it giving you Michael? -- -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [Asterisk-Users] oh323
All, I am trying PSTN Termination using Asterisk and SER. FXS---SER---Asterisk---FXO---PSTN I could configure FXS and SER. Do i need to register SER with Asterisk to forward call to Asterisk. Also, how to terminate there calls to respective FXO gateways. For achieving this, do i need to register FXO gateways with my Asterisk. regards, Nair.
[Asterisk-Users] Install Question
I have Wildcard TDM400P - so after compile i'm loading zapatel and wcfxs - kernel modules. My question is does I need other module in order to work with the FXO and FXS. ( beacause in other document i've noticed that there was writen to load and wcfxo - module ) but when i try this I get an error. voipgw:~# modprobe wcfxo /lib/modules/2.4.26/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.26/misc/wcfxo.o: insmod /lib/modules/2.4.26/misc/wcfxo.o failed /lib/modules/2.4.26/misc/wcfxo.o: insmod wcfxo failed ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: E1 yellow alarms
I had the exact same problem while installing a PRI yesterday. What I did was changing to letting the telco handle the timing( digit nr. 2 in span). However that didn't do the trick until I ran ztcfg -s and then ztcfg, and then the PRI resynced and has been fine since then. However it could be something else in your case, but I hope this helps someone. Regards, Maron Michiel Betel wrote: About every hour I see the yellow alarms on all or a number of channels of my PRI which is connected to the dutch telephony network, Asterisk keeps on working fine Here's an example where channel 1-24 went into alarm: WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 1: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 2: Yellow Alarm ... ... WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 23: Yellow Alarm WARNING[90124]: File chan_zap.c, Line 4139 (handle_init_event): Detected alarm on channel 24: No Alarm And right after that they are cleared again: NOTICE[90124]: File chan_zap.c, Line 4134 (handle_init_event): Alarm cleared on channel 1 NOTICE[90124]: File chan_zap.c, Line 4134 (handle_init_event): Alarm cleared on channel 2 ending with: WARNING[81931]: File chan_zap.c, Line 5137 (zt_pri_error): PRI: Read on 69 failed: Unknown error 500 Zaptel.conf has: # E1 card span=1,1,0,ccs,hdb3,crc4 # T1 card span=1,0,0,d4,ami # E1 bchan=1-15 dchan=16 bchan=17-31 # T1 fxoks=32-55 # Any ideas on how to get rid of these alarms? Thanks! Michiel Betel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS---SER---Asterisk---FXO---PSTN
All, I am trying PSTN Termination using Asterisk and SER. FXS---SER---Asterisk---FXO---PSTN I could configure FXS and SER. Do i need to register SER with Asterisk to forward call to Asterisk. Also, how to terminate there calls to respective FXO gateways. For achieving this, do i need to register FXO gateways with my Asterisk. regards, Nair.
[Asterisk-Users] making * more like a normal pbx
once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
[Asterisk-Users] TE410P in Austria
hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
On Mon, 2004-06-14 at 04:54, Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? so dial tone , 9, dialtone, then ur local num Look at ignorepat= -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Install Question
From what I have seen and read, the TDM400P does not require the wcfxo module, even though you have the FXO modules installed in it. It appears that the code to run the FXO modules has been rolled into the wcfxs module. Make sure you are running a fairly recent version of the zaptel source however. It is not in 0.7.2, but should be in .9 and newer CVS versions. Brian --- Damian Minkov [EMAIL PROTECTED] wrote: gt; I have Wildcard TDM400P - so after compile i'm gt; loading gt;zapatel and wcfxs - kernel modules. gt; gt; My question is does I need other module in order to gt; work with the FXO gt; and FXS. gt; ( beacause in other document i've noticed that there gt; was writen to load gt; and wcfxo - module ) gt; but when i try this I get an error. gt; gt;voipgw:~# modprobe wcfxo gt;/lib/modules/2.4.26/misc/wcfxo.o: init_module: No gt; such device gt;Hint: insmod errors can be caused by incorrect gt; module parameters, gt; including invalid IO or IRQ parameters. gt; You may find more information in syslog or gt; the output from dmesg gt;/lib/modules/2.4.26/misc/wcfxo.o: insmod gt; /lib/modules/2.4.26/misc/wcfxo.o failed gt;/lib/modules/2.4.26/misc/wcfxo.o: insmod wcfxo gt; failed gt; gt; ___ gt; Asterisk-Users mailing list gt; [EMAIL PROTECTED] gt; http://lists.digium.com/mailman/listinfo/asterisk-users gt; To UNSUBSCRIBE or update options visit: gt;http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
sorry, i don't know what you mean with: Is it the same as Telstra - could you please clearify this ? wolfgang Am Mo, den 14.06.2004 schrieb Michael M. Saunders um 12:25: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
How about cmd DISA ? Umar --- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-06-14 at 04:54, Jacob Hunter wrote: once u press 9 is there a way to make it so it restores dial tone,like most pbx's do? so dial tone , 9, dialtone, then ur local num Look at ignorepat= -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM AUDIOFiles
Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh Stream File or Get Data in my agi. It seems that there may be more than one GSM file type (with header and without, linear compressed, quadratically compressed--etc) I have read this doc-- but we need an answer which DOES NOT USE SOX: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk Can anyone please point me to, or post the correct way to produce a high quality, asterisk acceptable, GSM file WITHOUT using SOX? We will be using PROTOOLS-Mix-Plus, or can use PROTOOLS-HD to produce our audio files-- But are looking for ANY Macintosh or PC Audio-Production software suggestion , (outside of using SOX) which will take an .aiff or .wav file and turn it into a GSM file. ANY SUGGESTIONS WOULD BE GREATLY APPRECIATED- THANX. Thanks- JJQ _ Is your PC infected? Get a FREE online computer virus scan from McAfee® Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
Wolfgang, You need to use ccs not cas I would change your timing options to these:- -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 That should work At 12:33 14/06/2004 +0200, you wrote: sorry, i don't know what you mean with: Is it the same as Telstra - could you please clearify this ? wolfgang Am Mo, den 14.06.2004 schrieb Michael M. Saunders um 12:25: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice and DTMF
Does this work for every. If so I will add it to our knowledge base, so let me know. -James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Monday, June 14, 2004 3:32 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF It's official, Greg figured it out. And you know what, it all makes sense now: The scope for the dtmfmode setting is the section. Since the [broadvoice] section is needed for outgoing calls only, the [general] section -- the one containing the register directives would have to be where you define the dtmfmode for incoming connection. How about -- [general] dtmfmode=inband register = usera:[EMAIL PROTECTED] dtmfmode=rfc2833 register = userb:[EMAIL PROTECTED] Would that work? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marty Mastera Sent: Sunday, June 13, 2004 5:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF Greg, Per your suggestion, I added dtmfmode=inband to the general section of my sip.confthe other items you mentioned were already in sync with what I had. With that one change inbound DTMF to * IVR works! I will continue to play with it to flesh out it's reliability, but I was successfully able to navigate my IVR and log on to * VM. Thanks for the suggestion, I will followup with any interesting developments from my testing. Marty -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill Sent: Sunday, June 13, 2004 4:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Broadvoice and DTMF On Sat, 12 Jun 2004, Jay Milk wrote: Makes me think that the problem isn't with Broadvoice at all, but rather with Asterisk's DTMF recognition. I'm running CVS Head from late April. I'm running CVS-HEAD-06/06/04. I've spent a couple hours tinkering and taking notes on the dtmf issue this morning. I tried various combinations of rfc2833 and inband in my dtmfmode= statements in sip.conf and with each combination tried dialling out (xten softphone - * - BV - cell phone voicemail) and calling in (cell phone - BV - * - IVR) to test DTMF functionality (or brokenness). During each call, I used show channel in the CLI to see how * really thought the channel was configured. I think I finally came up with a setup where DTMF works. I'm hoping maybe some of you who have been struggling with this issue also will give it a try and tell the rest of us if it works in your config. sip.conf: [general] ... dtmfmode=inband [broadvoice] type=peer ... dtmfmode=inband [xtenphone] type=peer ... dtmfmode=rfc2833 I don't have any allow/disallow statements for any codecs, although I'm thinking about bringing those into my puzzle soon.. Anyway, with sip.conf set up as I described above, I placed a call: xten - * - BV - cell phone and the DTMF was passed through so that I could interact with the voicemail system. In this call, * indicated: xten - * channel: codec=GSM, dtmfmode=rfc2833 * - BV channel: codec=ULAW, dtmfmode=inband When I tried to set the xtenphone to use dtmfmode=inband (in sip.conf), * filled my console with Unable to process inband DTMF on 2 frames and I couldn't capture any info on the channel setup through the CLI. Removing the dtmfmode=inband statement under [general] didn't affect results. So then I placed another call: cell phone - BV - * I set up extensions.conf so that the incoming call from BV would go into an IVR I built for controlling xmms. I was able to enter the extension numbers to control the system. I don't have voicemail set up on this * box, so I couldn't test a call to that app. In this call * indicated: BV - * channel: codec=ULAW, dtmfmode=inband Removing the dtmfmode=inband statement under [general] DID affect results! With this statement commented out, * indicates: BV - * channel: codec=ULAW, dtmfmode=rfc2833 In this setup, DTMF broke and I couldn't control my xmms. So.. Jay, Michael, others.. if you try this config, let us know what results you find! Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 -
[Asterisk-Users] Re: GSM AUDIOFiles
In article [EMAIL PROTECTED], jeff quade [EMAIL PROTECTED] wrote: Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh Stream File or Get Data in my agi. It seems that there may be more than one GSM file type (with header and without, linear compressed, quadratically compressed--etc) I have read this doc-- but we need an answer which DOES NOT USE SOX: Why don't you want to use sox? I see from http://sox.sourceforge.net/ that it is available for Windows, and I would expect that as it compiles for BSD it would also compile for Mac OSX. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival application: clipping start of sound?
Hi I'm running a bright shiny new asterisk installation, and have discovered a problem with the festival application - when it plays back the generated sound, it skips the start. If, on the other hand, it has caching turned on, then when it plays the cached sound, it doesn't skip the first word or two. I assume that this has something to do with the time taken to generate the speech - is there anything I can do about this, apart from getting a faster machine for festival? Also, files in the festival cache directory seem to be created with mode . Is there any setting I need to prod to make them readable by asterisk? I'm running the debian packaged asterisk. thanks donald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] oh323
Does anyone have any ideas why this is failing -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael M. Saunders Sent: Monday, 14 June 2004 6:30 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] oh323 debian:/usr/src/asterisk-oh323-0.6.2# make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1385: too few arguments to function `ast_queue_control' chan_oh323.c: In function `oh323_hangup': chan_oh323.c:1417: too few arguments to function `ast_queue_hangup' chan_oh323.c: In function `oh323_read': chan_oh323.c:1855: too few arguments to function `ast_dsp_process' make[1]: *** [chan_oh323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' make: *** [subdirs_all] Error 1 debian:/usr/src/asterisk-oh323-0.6.2# -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stuart Grimshaw Sent: Monday, 14 June 2004 6:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 On Mon, 14 Jun 2004 18:06:25 +1000, Michael M. Saunders [EMAIL PROTECTED] wrote: This module wont compile can anyone give me any assistance Sure, what error messages is it giving you Michael? -- -S ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM AUDIOFiles
jeff quade wrote: I have read this doc-- but we need an answer which DOES NOT USE SOX: can't you acquire the audio files in .wav format? you can then: sox input.wav -r 8000 output.gsm polyphase ...on your linux box. We will be using PROTOOLS-Mix-Plus, or can use PROTOOLS-HD to produce our audio files-- But are looking for ANY Macintosh or PC Audio-Production software suggestion , (outside of using SOX) which will take an .aiff or .wav file and turn it into a GSM file. I still don't understand why you would so desperately need to do the conversion on a mac... anyway, I believe you might be able to compile sox on MacOS X Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM AUDIOFiles
jeff quade wrote: Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh Stream File or Get Data in my agi. It seems that there may be more than one GSM file type (with header and without, linear compressed, quadratically compressed--etc) Files edited in CoolEdit and saved as gsm, even with the proposed settings don't work for me too. Saved as .wav and converted with sox does the job. jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] collaboration with Panasonic PBX
If you mean connecting the X100P to an analog extension line then that will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. I can live without callerid for now. All I want to know is will it work. Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Both these options require that I'd put a matching interface in the PBX. This means purchasing a card for the PBX, for no small fee, and is undoable as there's no expansion room left in it. I have a couple of Analog line extensions free, and we thought to use them so we can make calls to our remote office via VOIP. Like so: |--- HQ -| |-- Remote Office -| Phone Extensions - PBX - Asterisk --- IP Phone The IP phones will be at the remote office. And users in HQ will just use the regular extensions, and dial 8 (for instance) as a prefix to get one of the two lines connected to Asterisk from the PBX. My question is can I use X100P cards to connect Analog lines from the PBX to Asterisk, and utilize both calls from HQ to the IP Phones and calls from the IP Phones to HQ this way. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as MGCP endpoint
It looks like Asterisk's mgcp, defaults as connect to endpoints or gateways. Is there a means to have it act as the endpoint or gateway? I have another system that needs to connect to a mgcp endpoint and I would like that to be asterisk. Thanks! Philip Kubat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] collaboration with Panasonic PBX
On Mon, 14 Jun 2004, Shoval Tomer wrote: Both these options require that I'd put a matching interface in the PBX. This means purchasing a card for the PBX, for no small fee, and is undoable as there's no expansion room left in it. Are all the isdn bri slots on the mainboard used already? Or can you get it with a mainboard without 4 bri ports? That was the only option when we purchased ours, but that was 7-8 years ago. The IP phones will be at the remote office. And users in HQ will just use the regular extensions, and dial 8 (for instance) as a prefix to get one of the two lines connected to Asterisk from the PBX. 8 is reserved for trunk-seiziure (followed by the number of the trunk to seize) by default on a kx-td1232. My question is can I use X100P cards to connect Analog lines from the PBX to Asterisk, and utilize both calls from HQ to the IP Phones and calls from the IP Phones to HQ this way. It ought to work. You will not get DDI support but that is not needed in your case I think. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
I think you'll find that Telsta is an Australian telco, not Austrian! Only a few miles out :-) -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Mon, 14 Jun 2004, Wolfgang Pichler wrote: sorry, i don't know what you mean with: Is it the same as Telstra - could you please clearify this ? wolfgang Am Mo, den 14.06.2004 schrieb Michael M. Saunders um 12:25: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
i've changed it now - but i still get the yellow alarm (i've changed it to: span=1,1,0,ccs,hdb3,crc4,yellow) this is my dmesg output after ztcfg -v - Found TE410P at base address 4010, remapped to d8b15000 TE410P version c01a009b FALC version: 0005, Board ID: 00 Reg 0: 0x124b1800 Reg 1: 0x124b1000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 9 (Austria) TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source TE410P: Span 2 configured for CCS/HDB3/CRC4 SPAN 2: Secondary Sync Source wct4xxp: Setting yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 2 --- any other ideas ? best regards Wolfgang Am Mo, den 14.06.2004 schrieb Jason Williams um 12:44: Wolfgang, You need to use ccs not cas I would change your timing options to these:- -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 That should work At 12:33 14/06/2004 +0200, you wrote: sorry, i don't know what you mean with: Is it the same as Telstra - could you please clearify this ? wolfgang Am Mo, den 14.06.2004 schrieb Michael M. Saunders um 12:25: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Welltech FXO: initial tests
Hi, I'm using the Welltech pstn GW 3804 (four analogue ports) and in some way I agree with Jorge's points. I am also using two Welltech SIP Phone LAN 201 I set them in proxy mode. I am still left with some problems. I can talk between the two SIP phones only with reinvite (I cannot talk when * stays in the middle) I can call the outside pstn line through the GW, but I cannot hear the ringing tone (from the caller) and cannot speak. When I call from pstn, the gateway answer after the specified number of rings but it does not forward the call to the lan phone extension. I set the GW in peer to peer mode. I will attach the * config files and the welltech phone and gw configuration if needed. Any help is really appreciated. Claudio Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to SIP. - We have tested only one port, it works well, audio quality is good (alaw). - Outgoing and incoming calls are working ok. - The Caller ID (from PSTN side) does not work - Answer supervision (reversal polarity detection) seems to work fine. This feature is very important to us, is the first time that we found this feature in a analog CO trunk. In a test application where we play a voice message to the called user, the message start to play just after answer. Tested with wire phone and cell phones. - Disconnect tone seems reliable (although the default configuration was not adjusted). We have done dozen of test in order to get the gw working. During the tests two issues came up, they need further analysis and tests: - Two times a UDP packages loop between the gw and * saturated the bandwidth after a hung up. Rebooting the gw does not stop the loop. Even with the gw turn off, * was sending the packages.Only rebooting * turn the system normal. - The gw port stay locked after a hung up. Apparently due to a no detection of the disconnect tone (in this case the tests were carried out with a PABX without disconnect tone). But the * user (SIP) was hung up and it seems that there are not a release timer. We will continue the tests and test the Welltech technical support as well (no required until now). Jorge -- Claudio Loletti (Lollo) mailto:[EMAIL PROTECTED] jid: [EMAIL PROTECTED] yahoo: [EMAIL PROTECTED] ICQ: 12096475 msn: [EMAIL PROTECTED] GnuPG public key available on keyservers Key fingerprint = 40AB B2CB 5022 507B 5167 587C B1BA 90AC 6ECD 94D9 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM Audio Files
Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people touch the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone else in the loop-- resampling the audio engineers .wav or .aiff files (hes a radio guy who works in .aiff at 44.1-32bit float) Im looking for a solution (software, and prefs) which will take the middle man (and SOX) out of the production loop-- ie the Audio Engineer simply masters and hands off GSM files which will work. I was hoping to find someone who has produced the correct gsm files without SOX, on either a MAC or PC. HELP, again, WOULD BE ***GREATLY APPRECIATED*** Cheers- jjq _ Looking to buy a house? Get informed with the Home Buying Guide from MSN House Home. http://coldwellbanker.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] where can I get toll-free number?
Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Audio Files
On 14 Jun 2004, at 14:17, jeff quade wrote: I was hoping to find someone who has produced the correct gsm files without SOX, on either a MAC or PC. http://www.versiontracker.com/dyn/moreinfo/macosx/18047 works just fine. Stephan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] collaboration with Panasonic PBX
Are all the isdn bri slots on the mainboard used already? Or can you get it with a mainboard without 4 bri ports? That was the only option when we purchased ours, but that was 7-8 years ago. We had a BRI Interface in the past, but we replace it with a 16 FXS board. Now the PBX is maxed out (8 lines, 48 FXS ports, plus 8 or 16 SMART FXS ports) The IP phones will be at the remote office. And users in HQ will just use the regular extensions, and dial 8 (for instance) as a prefix to get one of the two lines connected to Asterisk from the PBX. 8 is reserved for trunk-seiziure (followed by the number of the trunk to seize) by default on a kx-td1232. We have 12 trunk-seizures - if you're referring to CO lines. My question is can I use X100P cards to connect Analog lines from the PBX to Asterisk, and utilize both calls from HQ to the IP Phones and calls from the IP Phones to HQ this way. It ought to work. You will not get DDI support but that is not needed in your case I think. If DDI stands for Direct Dial In, then you're right, and I don't need them. Thanks for you help. Shoval ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] collaboration with Panasonic PBX
On Mon, 14 Jun 2004, Shoval Tomer wrote: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? I have an * under a Panasonic KX-TD816, as an extension for Panasonic, handling both incoming and outgoing calls. Caller-id is lost, as X100P is not a digital panasonic phone. I'm now moving to a different layout: asterisk is the PBX, routing calls to the Panasonic using a Cisco ATA 188 box. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival application: clipping start of sound?
IMHO the Festival application is slightly broken since it doesn't interface to the asterisk playback routines in a standard way. I've never had much luck with caching but have experienced the problem you outline on direct text conversions. This issue has been discussed on the bug tracker and this list in the past. You can hack Festival to pad out the pokayback with silence so the silence gets chopped before your sound. You can also have Festival save the sound file and then play back the sound using asterisk's standard playback routines. Both work but they're not nice solutions and add some latency, Iain --On Monday, June 14, 2004 10:58 pm +1200 Donald Gordon [EMAIL PROTECTED] wrote: Hi I'm running a bright shiny new asterisk installation, and have discovered a problem with the festival application - when it plays back the generated sound, it skips the start. If, on the other hand, it has caching turned on, then when it plays the cached sound, it doesn't skip the first word or two. I assume that this has something to do with the time taken to generate the speech - is there anything I can do about this, apart from getting a faster machine for festival? Also, files in the festival cache directory seem to be created with mode . Is there any setting I need to prod to make them readable by asterisk? I'm running the debian packaged asterisk. thanks donald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canadian DID
Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that would be even better! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prepaid application error
Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate(unknown, unknown) does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa' What is wrong ? Bye
RE: [Asterisk-Users] Broadvoice and DTMF
On Mon, 14 Jun 2004, Jay Milk wrote: It's official, Greg figured it out. And you know what, it all makes sense now: The scope for the dtmfmode setting is the section. Since the [broadvoice] section is needed for outgoing calls only, the [general] section -- the one containing the register directives would have to be where you define the dtmfmode for incoming connection. How about -- [general] dtmfmode=inband register = usera:[EMAIL PROTECTED] dtmfmode=rfc2833 register = userb:[EMAIL PROTECTED] Would that work? I haven't got time to test it this morning.. gotta run out to work. Unfortunately, work isn't playing with asterisk. Not yet. I think you're right, though.. my [broadvoice] section says type=peer. I wonder (but haven't tested) if using type=friend or adding a second section for type=user would have the same effect. BV would be considered our user when * receives a call from BV, so it makes sense that it might work to set the option there also. I'll try it this evening if nobody else has beaten me to it. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Audio Files
jeff quade wrote: The SOX resampled files work on our asterisk box-- but I gotta put someone else in the loop-- resampling the audio engineers .wav or .aiff files (hes a radio guy who works in .aiff at 44.1-32bit float) Im looking for a solution (software, and prefs) which will take the middle man (and SOX) out of the production loop-- ie the Audio Engineer simply masters and hands off GSM files which will work. I was hoping to find someone who has produced the correct gsm files without SOX, on either a MAC or PC. An alternative: export to WAV file, 8kHz 16bit integer. asterisk's quite happy with these, too (and you get a quality boost in no-compression channels). That's what I do, actually Apollon Koutlides ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepaid application error
hi, you have to also install the postgresql function's - the are included. There also exists a mailling list especialy for the prepaid application - take a look at asteriskbilling @ sourceforge best regards Wolfgang Am Mo, den 14.06.2004 schrieb [EMAIL PROTECTED] um 14:58: Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate(unknown, unknown) does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa' What is wrong ? Bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTERISK V. SER
Hi Guys, Can somone explain differances between SER and ASTERISK. I am particularly interested in functionality that is not available with ASTERISK but SER can provide. Best Regards Mit freundlichen Grüssen Meilleures Salutations med vennlig hilsen Reza Kordi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK V. SER
Hello, I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution combining multiple protocols (IAX, H323, SIP, Skinny, MGCP, SCCP) so that they can each talk to eachother and multiple codecs (one can use G729 and the other can use ULAW for example). Asterisk also provides other features such as voicemail, hold on music, call display, etc. - Joshua Colp. - Original Message - From: Reza Kordi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 10:55 AM Subject: [Asterisk-Users] ASTERISK V. SER Hi Guys, Can somone explain differances between SER and ASTERISK. I am particularly interested in functionality that is not available with ASTERISK but SER can provide. Best Regards Mit freundlichen Grüssen Meilleures Salutations med vennlig hilsen Reza Kordi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTERISK V. SER
Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dyn Exten
That is a nice pseudo dynamic solution. But is asterisk planning to have dynamic extensions support?? Jeremy McNamara wrote: Pablo Endres wrote: For eficiency, I create a temp file, and diff from the previous version (so I don't reload if I don't have to). Why bother doing that much processing? Just set a flag somewhere that determines weather or not you need a reload. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK V. SER
Well since a person would normally go for usability and asterisk is originally created for Linux, I said it was a Linux PBX Solution. I have nothing against BSD myself, I have a FreeBSD sitting a few feet away from me. - Joshua Colp. - Original Message - From: Aaron J. Angel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:24 AM Subject: RE: [Asterisk-Users] ASTERISK V. SER Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
Wouldn't this keep the 9 since you didn't use {EXTEN}:1 to include the
StripMSD portion?
I found this thread interesting, as it apears simpler than the dialplan
I used:
exten = _9NXX,1,StripMSD,1
exten = _NXX,2,Prefix,1512
exten = _1512NXX,3,Dial(${TRUNK1}/${EXTEN})
exten = _1512NXX,4,Dial(${TRUNK2}/${EXTEN})
exten = _1512NXX,5,Congestion
usedcanon wrote:
looks fine to me
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 12:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
Does this look right
exten = _9NXX,1,SetCallerID(831-XXX-)
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten = _9NXX,3,Congestion
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[Asterisk-Users] prepaid running mysql
any prepaid app running in MYSQL? I already have mysql and dont want to add postgres.. thanks anybody. H.C ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_Capi 0.3.4
Just tried compiling chan_capi 0.3.4 under CVS Head and get the following errors. chan_capi.c:60: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:61: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:62: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:63: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:64: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:85: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_capi.c:86: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [chan_capi.o] Error 1 version 0.3.3 has been running fine without issues Can any one assist (I'm sticking with 0.3.3 for now) Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 'background' problem
On Sat, 2004-06-12 at 17:47, Mark Elkins wrote: I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. Let me try again... If I mix background announcements with SayUnixTime - then my IVR menu system breaks - DTMF tones are not recognised. Is this a Bug? What is the work around? My example was... exten = s,7,Playback(posix-welcome-afterhours) ; Welcome to Posix; Systems After hours support, Our business hours are Monday ; to Friday, 8am to 5pm. The time is now exten = s,8,SayUnixTime(||AIMP); A:Day, I:Hours, M:Minutes, P:am/pm exten = s,9,Playback(posix-welcome-afterhours-try) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P in Austria
hi all, i have now enabled pri debugging with pri intense debug span so i get: Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Urgent handler Sending Set Asynchronous Balanced Mode Extended what does this all means ? best regards Wolfgang Am Mo, den 14.06.2004 schrieb Wolfgang Pichler um 14:06: i've changed it now - but i still get the yellow alarm (i've changed it to: span=1,1,0,ccs,hdb3,crc4,yellow) this is my dmesg output after ztcfg -v - Found TE410P at base address 4010, remapped to d8b15000 TE410P version c01a009b FALC version: 0005, Board ID: 00 Reg 0: 0x124b1800 Reg 1: 0x124b1000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 9 (Austria) TE410P: Span 1 configured for CCS/HDB3/CRC4 SPAN 1: Primary Sync Source TE410P: Span 2 configured for CCS/HDB3/CRC4 SPAN 2: Secondary Sync Source wct4xxp: Setting yellow alarm on span 1 wct4xxp: Setting yellow alarm on span 2 --- any other ideas ? best regards Wolfgang Am Mo, den 14.06.2004 schrieb Jason Williams um 12:44: Wolfgang, You need to use ccs not cas I would change your timing options to these:- -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 That should work At 12:33 14/06/2004 +0200, you wrote: sorry, i don't know what you mean with: Is it the same as Telstra - could you please clearify this ? wolfgang Am Mo, den 14.06.2004 schrieb Michael M. Saunders um 12:25: Is it the same as Telstra -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang Pichler Sent: Monday, 14 June 2004 8:11 PM To: Asterisk-Users Mailinglist Subject: [Asterisk-Users] TE410P in Austria hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another technics told me that it could be that i will need a different cable - could this be ? At time my configuration is: -/etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 span=2,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 bchan=32-46 bchan=48-62 dchan=16 dchan=47 loadzone=at defaultzone=at ---/etc/asterisk/zaptel.conf- [channels] switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 1 context = default channel = 1-15 channel = 17-31 switchtype = euroisdn signalling = pri_cpe pridialplan = local group = 2 context=default channel = 32-46 channel = 48-62 Another thing - for the wct4xxp module - do i really need to restart the whole pc so that changes in /etc/zaptel.conf take effect, or do i only need to reinsert the module - rerun ztcfg ? hope someone here can help me best regards Wolfgang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] collaboration with Panasonic PBX
Peter Svensson wrote: On Mon, 14 Jun 2004, Shoval Tomer wrote: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? If you mean connecting the X100P to an analog extension line then that will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. That's a shame- what protocol do they use? DTMF? http://voip-info.org/wiki-Asterisk+bounty+non-Bellcore-CLID Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Do you use the TD-1232's 'T1' interface, then? - with what PRI card? Digium or Cisco or what? - does it support Q.931? Their webpage is vague as to exactly what they mean by 'T1' http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=FstoreId=11251catalogId=11005itemId=62983catGroupId=2modelNo=KX-TD1232surfModel=KX-TD1232ignoreRedirect=1 Thanks for any extra information - I need to interface * with one of these in 2 locations. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 - Adtran and SIP
Hello, I have installed T1 card and Adtran TSU 600. Everything works ok but... When I am making local (in our local network) calls using FXS ports with Adtran ,connected with asterisk using T1 card, to a SIP phone also in our local network everything works very good. Problems start when I am making calls using Adtran outside our local network (for example internet) to SIP destination. The voice is getting choppy... But when I am making call using SIP phone to the same destination call is OK. What could be the problem ? Regards, bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] If IAX client is not logged in/registered, Dial plan executes BUSY vs UNAVAILABLE
If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's I'm on the phone at the moment message vs. the I'm unavailable message. Is this by design? Here's the extension in question's dialplan: ;extensions.conf exten = 106,1,Dial(IAX2/nikko,20,tT) exten = 106,2,Voicemail(u105) exten = 106,3,Hangup exten = 106,102,Voicemail(b105) exten = 106,103,Hangup And here's the CLI debug: pbxMobile*CLI -- Executing Dial(SIP/nmartin-aeca, IAX2/nikko|20|tT) in new stack pbxMobile*CLI Jun 14 10:28:26 NOTICE[4997140]: app_dial.c:554 dial_exec: Unable to create channel of type 'IAX2' pbxMobile*CLI == Everyone is busy at this time pbxMobile*CLI -- Executing VoiceMail(SIP/nmartin-aeca, b105) in new stack pbxMobile*CLI -- Playing 'voicemail/default/105/busy' (language 'en') pbxMobile*CLI -- Playing 'vm-intro' (language 'en') pbxMobile*CLI == Spawn extension (Outgoing, 106, 102) exited non-zero on 'SIP/nmartin-aeca' pbxMobile*CLI -- Executing Hangup(SIP/nmartin-aeca, ) in new stack pbxMobile*CLI == Spawn extension (Outgoing, h, 1) exited non-zero on 'SIP/nmartin-aeca' pbxMobile*CLI exit Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] collaboration with Panasonic PBX
We also are working to interface a Panasonic pbx with a pri. Calls from * to Panasonic via pri suport caller id and DID. However, I have not found a way to forward calls out the pri without manually dialing the trunk group for the pri and then the extension number. I have overlap dialing enabled in the panasonic. Any tips to make say a specific extension able to forward to the pri with destination digits getting sent? Peter Svensson wrote: On Mon, 14 Jun 2004, Shoval Tomer wrote: Here's my question: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? If you mean connecting the X100P to an analog extension line then that will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- respectfully, Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Canadian DID
DID's from Allstream (ATT) are $2 Cdn/month but I think they have a rule that it has to terminate on their network somewhere... -Original Message- From: Linus Surguy [mailto:[EMAIL PROTECTED] Sent: Monday, June 14, 2004 6:53 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that would be even better! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] making * more like a normal pbx
Title: Message You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program, tell them jacob, hunter81 sent you
RE: [Asterisk-Users] where can I get toll-free number?
I'm using zoneld numbers which I can terminate on any US number -- http://ld.net/mu has various options. You basically get your incoming voicepulse, broadvoice, etc line, then get an 800# to terminate on those lines and you're in asterisk. Through this, I also have tollfree numbers to my cellphones and fax... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Hajek Sent: Monday, June 14, 2004 7:30 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] where can I get toll-free number? Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk real life examples and case studies ?
I can't seem to find the link to examples of asterisk installations for different sized sites. I'm not after specific configuration of the conf files, just an overview on what hardware/chassis cards people are running and what channels - phones etc people are using. Cheers * newbie peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] collaboration with Panasonic PBX
On Mon, 14 Jun 2004, Fran Boon wrote: Peter Svensson wrote: will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. That's a shame- what protocol do they use? DTMF? http://voip-info.org/wiki-Asterisk+bounty+non-Bellcore-CLID Analog extensions is what I ment. Sorry for the confusion. There is no callerid at all on the analog estensions, only on the digital (system phones) extensions and the internal isdn busses. There is an add-on card (KX-TD193) that provieds caller id for 4 extensions. Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Do you use the TD-1232's 'T1' interface, then? - with what PRI card? Digium or Cisco or what? - does it support Q.931? No, their E1 PRI card. I think they are not the same. There was a 1 digit difference in the model number. We connect it to a TE405P which is also connected to the pstn. All isdn lines use EuroISDN (q.931 with etsi modifications). Their webpage is vague as to exactly what they mean by 'T1' http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=FstoreId=11251catalogId=11005itemId=62983catGroupId=2modelNo=KX-TD1232surfModel=KX-TD1232ignoreRedirect=1 They have a lot of different interface cards for the kx-td1232. KX-TD290 is a E1 PRI card ccording to our documentation. However, from reading on the web it may also be a T1 PRI This is strange, you should contact your sales representative lest you end up with a doorstop. KX-TD187 T1 without isdn Thanks for any extra information - I need to interface * with one of these in 2 locations. You can contact me privatly if you want more information. Peter -- Peter Svensson ! Pgp key available by finger, fingerprint: [EMAIL PROTECTED]! 8A E9 20 98 C1 FF 43 E3 07 FD B9 0A 80 72 70 AF Remember, Luke, your source will be with you... always... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom IP 600
I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if anyone can confirm that this works with the polycoms. I know the 7960s support this, but I want to make sure the Polycom sales team wasn't just saying Yes to make the sale. Any comments are appreciated. -Eric -Original Message- Subject: fwd on busy when calling multiple extensions at once Chris A. Icide wrote: IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I base this off of having had both an IP600 and a 7960. The two advantages the 7960 had over the IP600 was appearance and ease of configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down. Now, you MAY want to try registering all 6 lines on the polycom to the same line and see if the phone handles that as well as the cisco. If it does, then you are set. Otherwise, you will need some complex configuration work in your extensions.conf to achieve what you are looking to achieve. Some thoughts: What do you want to happen when one of the call takers has all 6 lines in use? Have you considered using queues to do what you need? -Chris On 10:08 AM 5/22/2004, Brian Cuthie wrote: You might consider using the Cisco SIP phones. They're smart enough to accept incoming calls for as many call appearances you have with the same SIP registration. -brian Tor Roberts wrote: Hi, I am setting up a dispatch center where will have 4 call takers, all with Polycom IP 600 Sip phones. Each phone will be setup with 6 extensions each. When a new call comes in, the first extension on all the phones will ring. This works fine, the problem is when one of the dispatchers is already using her first extension and another call comes in. What happens now is that the remaining 3 phones ring on the first extension, but the dispatcher who is on a call, her phone does not ring. I want her second extension ring along with the other 3 phones first extensions. In sip.conf I have all the extensions set to incominglimit=1 and the pertinent part of extensions.conf is: exten = s,1,Dial(SIP/5000SIP5001SIP5002SIP5003,20,tr) exten = s,2,Dial(SIP/5004SIP5005SIP5006SIP5007,20,tr) and so on. If anybody has any insight, or a better solution, that would be great. Thanks, -Tor Roberts ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepaid application error
Dear List I have try to install the app_prepaid but after i compile it with no problem i start * (cvs branch) and say this error: - undefined symbol: PQexec Can someone give some tips? Thanks in advance Dimitri On Monday 14 June 2004 02:58 pm, [EMAIL PROTECTED] wrote: Hi, I successfully installed postgres and prepaid application in my asterisk box but after I digited the code I receive this error: ERROR: Function asterisk_authenticate(unknown, unknown) does not exist Unable to identify a function that satisfies the given argument types You may need to add explicit typecasts -- Playing 'prepaid-no-aaa' What is wrong ? Bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Audio Files
jeff == jeff quade [EMAIL PROTECTED] writes: jeff Im looking for a solution (software, and prefs) which will take jeff the middle man (and SOX) out of the production loop-- ie the jeff Audio Engineer simply masters and hands off GSM files Unless you are using the gsm codec close to exclusively on the wire, there is no important reason to use it for the on-disk files. Try to get your AE to save them as mono 16bit signed-linear wav files. He should understand that and be able to do so. This will save the decode from gsm step when playing those files. It will also slightly improve the quality when compressing to some other codec. Especially if it is one of the lower bandwidth codecs. (If you are only ever sending the files out over zap channels, the best file format to use is ulaw or alaw as applicable to your area. But that may be as difficult to get from your engineer as gsm.) -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area code for 7 digit dialing
ya mine worked.
--
Gafachi.com - referal code hunter81
instant iax termination - 2 cents a minute
Also they have a great referal program,
tell them jacob, hunter81 sent you
On Jun 14, 2004, at 7:31 AM, Charlie Hedlin wrote:
Wouldn't this keep the 9 since you didn't use {EXTEN}:1 to include the StripMSD portion?
I found this thread interesting, as it apears simpler than the dialplan I used:
exten => _9NXX,1,StripMSD,1
exten => _NXX,2,Prefix,1512
exten => _1512NXX,3,Dial(${TRUNK1}/${EXTEN})
exten => _1512NXX,4,Dial(${TRUNK2}/${EXTEN})
exten => _1512NXX,5,Congestion
usedcanon wrote:
looks fine to me
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacob Hunter
Sent: 12 June 2004 12:48
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Prepending for 9NxxNxxx - adding the area
code for 7 digit dialing
Does this look right
exten => _9NXX,1,SetCallerID(831-XXX-)
exten => _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/1831${EXTEN})
exten => _9NXX,3,Congestion
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Re: [Asterisk-Users] where can I get toll-free number?
On Jun 14, 2004, at 9:06 AM, Jay Milk wrote: I'm using zoneld numbers which I can terminate on any US number -- http://ld.net/mu has various options. You basically get your incoming voicepulse, broadvoice, etc line, then get an 800# to terminate on those lines and you're in asterisk. Through this, I also have tollfree numbers to my cellphones and fax... Yeah, but that's more expensive then NuFone's $0.029/minute 800 rate. If you're looking for an 800 number that points to an existing device, then ld.net probably a great way to go. If you're looking for 800 VoIP services, then there's no reason to stack services like this. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sayson IP Phones?
Michael Graves wrote: No, sir. Have not seen IP300. However, a friend loaned me a IP600 for evaluation. I have yet to figure out its support and configuration. Looks like a nice instrument. The Polycom SoundPoint IP300 is not an SIP-capable phone; H.323 and MGCP only. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: fax obsoleted? Was: Re: Fax via email
Hi. Just my $0.02 worth on this question. However, fax is still very much alive and healthy in the area of imaged document exchange where the website or e-mail use would not be appropriate - i.e., where the sender wants to initiate the document exchange and the document is in a more-than-text form or image of some kind (applications, completed applications, handwriting, etc.). Furthermore, I don't see this usage of fax going away any time soon. Indeed, technology seems to be providing better and better ways for this to continue, and I see no end to this sender-initiated use of fax. Agreed to some extent. However, most cases I think are somewhat overstated. I.e. people do fax applications back and forth in part because formats like MS Word don't necessarily guarantee layout. If PDF's were more commonly used, this would be more obsolete. But PDF producing and editing software is expensive, and you cannot exactly sign a PDF with your handwritten signature and send it back using standard hardware and software. On the other hand, there is a general perception that a signature on a fax will be somewhat better than a signature on an email from a legal perspective (IANAL, though). This is where I have seen the greatest continuation of the fax. It's worth pointing out that e-mail works off of a different premise than fax, and therefore cannot ever fully obsolete fax as it is. Unlike a website or fax, e-mail does not provide a mechanism for both the sender and the receiver to negotiate the communication and presentation of the document. E-mail permits the sender to send arbitrary filetypes with arbitrary formatting which the receiver may or may not be able to utilize easily. With a website the receiver should be aware of what they are clicking on, and with fax the receiver capabilities are communicated at the outset to the sender, and the sender must select tranmission parameters from those capabilites. Thus, with fax the sender can have a reasonably good degree of confidence that when the receiver sends the confirmation signal (MCF) that the receiver can view the document and that it appears to the receiver nearly exactly the same way as it appears to the sender. Not only can you not do this with e-mail, but furthermore with e-mail you only know that your outbound mail relay has accepted the mail or not. You do not have any reassurance that that the intended recipient actually did receive the message. And with the large amount of spam out there (very large in comparison to the quantity of junk faxes), spam filters, e-mail viruses, and such, e-mail really isn't a very good means to transmit these kinds of things. Fair enough. But I think the largest advantage I have seen you point out is the fact that email is packet-switched, and store-and-forward while the fax is connection-switched and delivered immediately or not at all. Therefore fax systems are fundamentally more reliable than email. But, let me ask you this. Suppose that I produced PDF forms which could be edited and then uploaded into a drop-box on my web site. This does not address some of the concerns about www/email vs fax, but it does address many of them. I suspect that you are right-- that fax services will eventually become merged with the internet, but we will need to see how that exactly occurs. Best Wishes, Chris Travers begin:vcard fn:Chris Travers n:Travers;Chris email;internet:[EMAIL PROTECTED] x-mozilla-html:FALSE version:2.1 end:vcard
Re: [Asterisk-Users] Asterisk Agent Logoff?
Aaron J. Angel wrote: What about logging out? Is it possible to have Asterisk log the agent out without having to specify the agent password and hitting # twice? Auto-logout would be nice to have too... if they don't answer a call directed to them, they wouldn't get any more until they log back in. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk real life examples and case studies ?
On Mon, 2004-06-14 at 11:13, Peter Mitchell wrote: I can't seem to find the link to examples of asterisk installations for different sized sites. I'm not after specific configuration of the conf files, just an overview on what hardware/chassis cards people are running and what channels - phones etc people are using. Go to the WiKi boy. www.voip-info.org -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] german localization for mailbox available?
hi, i just wanted to ask if there is a german localization for the audio files of the mailbox available on the net. regards frank sautter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk real life examples and case studies ?
Peter Mitchell wrote: I can't seem to find the link to examples of asterisk installations for different sized sites. I'm not after specific configuration of the conf files, just an overview on what hardware/chassis cards people are running and what channels - phones etc people are using. here is one from my bookmarks: http://graphics.cs.uni-sb.de/VoIP/en/index.html jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dyn Exten
Jose R. Ortiz Ubarri wrote: That is a nice pseudo dynamic solution. But is asterisk planning to have dynamic extensions support?? One can already do dynamic extensions with Asterisk TODAY. You have to be smarter than what your working on. Jeremy McNamara Jeremy McNamara wrote: Pablo Endres wrote: For eficiency, I create a temp file, and diff from the previous version (so I don't reload if I don't have to). Why bother doing that much processing? Just set a flag somewhere that determines weather or not you need a reload. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice and DTMF
At 04:39 PM 6/13/2004 -0600, you wrote: Greg, Per your suggestion, I added dtmfmode=inband to the general section of my sip.confthe other items you mentioned were already in sync with what I had. With that one change inbound DTMF to * IVR works! I will continue to play with it to flesh out it's reliability, but I was successfully able to navigate my IVR and log on to * VM. Thanks for the suggestion, I will followup with any interesting developments from my testing. Marty Yes, indeed: the addition of dtmfmode=inband above the registrations in sip.conf is the key. It's working for me, too. I worship your research skills, Greg! :-) Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] do_monitor warning message
Hi, I'm using Asterisk version (Asterisk CVS-05/30/04-16:28:04) on Debian Woody. Sometimes I get this warning message: Jun 14 13:32:41 WARNING[10251]: chan_zap.c:5044 do_monitor: select return -1: Bad file descriptor When that is happening, Asterisk gets slow and close all remote active connections (asterisk -vvvcr). VoIP call alse gets bad at this time. That message appears many times, like if Asterisk were in loop. I've downloaded the version from the CVS (cvs checkout -r 1-0_stable). And I'm not using any monitor application.. Connected to this server, I have a Zhone (16 FXS extensions and 8 FXO lines) and a T100P. I'm also using the last CVS version for Zaptel and Libpri. Did anybody get the same message? How can I fix that? regards Oz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
I searched for these things, however I don't know the proper terminology, so I come on here, people give me ideas, then i look on wiki. -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you On Jun 14, 2004, at 8:59 AM, Jay Milk wrote: You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
[Asterisk-Users] compile error with asterisk-addons
I try to install asterisk-addons but I get an error. Who can help me? Is my MySQL not complete or do I have another problem? [EMAIL PROTECTED] asterisk-addons]# make install cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 [EMAIL PROTECTED] asterisk-addons]# Best regards, Han van Hulst ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Number Portability and VoicePulse
I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nextel phone and mute on Asterisk?
Hello, I have a really irritating issue that I haven't had time to investigate much - I hope someone has encountered it and can tell me a solution. I didn't see anything in searching archives / sites.. When my Nextel i90c phone gets a page (2 way text message via the internet option) it has an irritating tone to get me to hear it. However this tone seems to mute asterisk (reproducible). Is there something I can configure to disable a mute? This on a Cisco ATA186 device definitely - I haven't been able to reproduce on other devices (haven't had time to investigate on other devices). Steve Radich BitShop, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] where can I get toll-free number?
Good deal, I didn't know about nufone's service -- but then, how could I, it's not on their site. I've been to www.nufone.net a few times, and looking at their site, you'd think you're dealing with a front-business for the mafia. If they'd publish rates and a lists of rate-centers, they could make good business. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Laird Sent: Monday, June 14, 2004 11:29 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] where can I get toll-free number? On Jun 14, 2004, at 9:06 AM, Jay Milk wrote: I'm using zoneld numbers which I can terminate on any US number -- http://ld.net/mu has various options. You basically get your incoming voicepulse, broadvoice, etc line, then get an 800# to terminate on those lines and you're in asterisk. Through this, I also have tollfree numbers to my cellphones and fax... Yeah, but that's more expensive then NuFone's $0.029/minute 800 rate. If you're looking for an 800 number that points to an existing device, then ld.net probably a great way to go. If you're looking for 800 VoIP services, then there's no reason to stack services like this. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: fax obsoleted? Was: Re: Fax via email
On 2004.06.14 09:31 Chris Travers wrote: But, let me ask you this. Suppose that I produced PDF forms which could be edited and then uploaded into a drop-box on my web site. This does not address some of the concerns about www/email vs fax, but it does address many of them. You creating a PDF, hanging it on your website, and then requiring the recipient to download the PDF does not constitute what I was referring to as sender intitiated communication. You're still requiring the recipient to take an active part in the communication. With fax, the participation of the recipient can be passive. So the scenario that you outline here is already web-only. We don't commonly see people asking receipients to poll faxes any more. But people are regularly instructed to visit websites. The non-obsoleted functionality of fax that I was describing was one where the sender initiates the communication and the recipient does nothing more but passive participation (i.e., has fax-receiving equipment on standby). Lee. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 2000 not answering em_w calls
I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I can not answer it. With Sip debug set I will see a bunch of messages but the following messages seems important: 7 headers, 18 lines Jun 14 11:35:08 WARNING[4101]: chan_sip.c:2012 find_call: Call missing call ID from '192.168.50.119' (192.168.50.119 is the IP of my Sipura device) Calls coming in from the phone company though em_w trunks work fine when terminated to analog phones (fxo_ks via a channel bank) or to another pbx analog trunks (fxs_ks via a channel bank) or to a Grandstream sip phone. So, I do not know whether this is a Sipura 2000 problem or an * problem. Does anyone have any light to shine on the subject. Asterisk CVS-HEAD-06/14/04-09:03:15 built by [EMAIL PROTECTED] on a i686 running Linux Sipura 2000 software version 1.0.33 Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
Title: Message I think he just wants to promote gafachi.com - Original Message - From: Jay Milk To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:59 AM Subject: RE: [Asterisk-Users] making * more like a normal pbx You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program, tell them jacob, hunter81 sent you
RE: [Asterisk-Users] Dyn Exten
Jeremy McNamara wrote: Jose R. Ortiz Ubarri wrote: That is a nice pseudo dynamic solution. But is asterisk planning to have dynamic extensions support?? One can already do dynamic extensions with Asterisk TODAY. You have to be smarter than what your working on. Jeremy, your response is hardly worth that time it took to type it. Please explain. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Number Portability and VoicePulse
I didn't think voicepulse does number portability.. I've been looking for an IAX provider that offers number portability, and I have yet to find one :( -Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James W. Brinkerhoff Sent: Monday, June 14, 2004 1:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Number Portability and VoicePulse I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk real life examples and case studies ?
Steven Critchfield wrote: On Mon, 2004-06-14 at 11:13, Peter Mitchell wrote: I can't seem to find the link to examples of asterisk installations for different sized sites. I'm not after specific configuration of the conf files, just an overview on what hardware/chassis cards people are running and what channels - phones etc people are using. Go to the WiKi boy. www.voip-info.org Digium recently added case stories, check http://www.digium.com /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
no, i have no affiliation with them. I just think they have great service. j hunter [EMAIL PROTECTED] On Jun 14, 2004, at 10:48 AM, Steve Totaro wrote: I think he just wants to promote gafachi.com x-tad-bigger- Original Message -/x-tad-bigger x-tad-bigger /x-tad-biggerx-tad-biggerFrom:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-biggerJay Milk/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerTo:/x-tad-biggerx-tad-bigger /x-tad-biggerx-tad-bigger[EMAIL PROTECTED]/x-tad-biggerx-tad-bigger /x-tad-bigger x-tad-biggerSent:/x-tad-biggerx-tad-bigger Monday, June 14, 2004 11:59 AM/x-tad-bigger x-tad-biggerSubject:/x-tad-biggerx-tad-bigger RE: [Asterisk-Users] making * more like a normal pbx/x-tad-bigger You really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: Monday, June 14, 2004 4:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] making * more like a normal pbx once u press 9 is there a way to make it so it restores dial tone, like most pbx's do? so dial tone , 9, dialtone, then ur local num -- Gafachi.com - referal code hunter81 instant iax termination - 2 cents a minute Also they have a great referal program, tell them jacob, hunter81 sent you
Re: [Asterisk-Users] where can I get toll-free number?
On Jun 14, 2004, at 10:40 AM, Jay Milk wrote: Good deal, I didn't know about nufone's service -- but then, how could I, it's not on their site. I've been to www.nufone.net a few times, and looking at their site, you'd think you're dealing with a front-business for the mafia. If they'd publish rates and a lists of rate-centers, they could make good business. Well, yeah. It's hard to argue with that :-). Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Number Portability and VoicePulse
Sounds like good questions to ask VoicePulse. - Original Message - From: James W. Brinkerhoff [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 1:14 PM Subject: [Asterisk-Users] Number Portability and VoicePulse I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 not answering em_w calls
Well, I don't think it's the sipura. We have 45 SPA-2000 adapters connected to 82 analog phones going through 3 asterisk servers all using EM Wink start T1s, and we have no problems with inbound or outbound on any of the sipura adapter-connected phones. Post your sip.conf entry for your sipura device, as well as what line features you have activated on your sipura configuration. MATT--- -Original Message- From: Don Pobanz [mailto:[EMAIL PROTECTED] Sent: Monday, June 14, 2004 1:41 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Sipura 2000 not answering em_w calls I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I can not answer it. With Sip debug set I will see a bunch of messages but the following messages seems important: 7 headers, 18 lines Jun 14 11:35:08 WARNING[4101]: chan_sip.c:2012 find_call: Call missing call ID from '192.168.50.119' (192.168.50.119 is the IP of my Sipura device) Calls coming in from the phone company though em_w trunks work fine when terminated to analog phones (fxo_ks via a channel bank) or to another pbx analog trunks (fxs_ks via a channel bank) or to a Grandstream sip phone. So, I do not know whether this is a Sipura 2000 problem or an * problem. Does anyone have any light to shine on the subject. Asterisk CVS-HEAD-06/14/04-09:03:15 built by [EMAIL PROTECTED] on a i686 running Linux Sipura 2000 software version 1.0.33 Don Pobanz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dyn Exten
On Mon, 2004-06-14 at 13:04, Aaron J. Angel wrote: Jeremy McNamara wrote: Jose R. Ortiz Ubarri wrote: That is a nice pseudo dynamic solution. But is asterisk planning to have dynamic extensions support?? One can already do dynamic extensions with Asterisk TODAY. You have to be smarter than what your working on. Jeremy, your response is hardly worth that time it took to type it. Please explain. There are many ways to get it done right now, any number of methods apply. If you need it you should be able to figure it out. That is where Jeremy was probably heading. It basically is a general suggestion to open the mind not the email client and think about it for a while. You will need to decide on the best route for you and your setup. Now you have made me waste more time telling you to think about the problem a bit more intelligently, and I still haven't hand held you to the answer. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Welltech FXO: initial tests
Claudio, Try the SIP version 103. It is rock solid, and have FSK CID. Unfortunately, I think, DTMF CID is used in Italy, which is not (yet?) supported. Also, polarity detection is missing in this version, but Welltech promise to be included in the next release. However, your problems seems to be a bad gw configuration. Jorge Claudio Loletti wrote: Hi, I'm using the Welltech pstn GW 3804 (four analogue ports) and in some way I agree with Jorge's points. I am also using two Welltech SIP Phone LAN 201 I set them in proxy mode. I am still left with some problems. I can talk between the two SIP phones only with reinvite (I cannot talk when * stays in the middle) I can call the outside pstn line through the GW, but I cannot hear the ringing tone (from the caller) and cannot speak. When I call from pstn, the gateway answer after the specified number of rings but it does not forward the call to the lan phone extension. I set the GW in peer to peer mode. I will attach the * config files and the welltech phone and gw configuration if needed. Any help is really appreciated. Claudio Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to SIP. - We have tested only one port, it works well, audio quality is good (alaw). - Outgoing and incoming calls are working ok. - The Caller ID (from PSTN side) does not work - Answer supervision (reversal polarity detection) seems to work fine. This feature is very important to us, is the first time that we found this feature in a analog CO trunk. In a test application where we play a voice message to the called user, the message start to play just after answer. Tested with wire phone and cell phones. - Disconnect tone seems reliable (although the default configuration was not adjusted). We have done dozen of test in order to get the gw working. During the tests two issues came up, they need further analysis and tests: - Two times a UDP packages loop between the gw and * saturated the bandwidth after a hung up. Rebooting the gw does not stop the loop. Even with the gw turn off, * was sending the packages.Only rebooting * turn the system normal. - The gw port stay locked after a hung up. Apparently due to a no detection of the disconnect tone (in this case the tests were carried out with a PABX without disconnect tone). But the * user (SIP) was hung up and it seems that there are not a release timer. We will continue the tests and test the Welltech technical support as well (no required until now). Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dyn Exten
Best mailling list support I've ever read!!! Thanks a lot for your help. Steven Critchfield wrote: On Mon, 2004-06-14 at 13:04, Aaron J. Angel wrote: Jeremy McNamara wrote: Jose R. Ortiz Ubarri wrote: That is a nice pseudo dynamic solution. But is asterisk planning to have dynamic extensions support?? One can already do dynamic extensions with Asterisk TODAY. You have to be smarter than what your working on. Jeremy, your response is hardly worth that time it took to type it. Please explain. There are many ways to get it done right now, any number of methods apply. If you need it you should be able to figure it out. That is where Jeremy was probably heading. It basically is a general suggestion to open the mind not the email client and think about it for a while. You will need to decide on the best route for you and your setup. Now you have made me waste more time telling you to think about the problem a bit more intelligently, and I still haven't hand held you to the answer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making * more like a normal pbx
ignore pat 9 - Original Message - From: Jacob Hunter To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 2:16 PM Subject: Re: [Asterisk-Users] making * more like a normal pbx no, i have no affiliation with them. I just think they have great service.j hunter[EMAIL PROTECTED]On Jun 14, 2004, at 10:48 AM, Steve Totaro wrote: I think he just wants to promote gafachi.com- Original Message -From: Jay Milk To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:59 AMSubject: RE: [Asterisk-Users] making * more like a normal pbxYou really need to start making friends with google and the wiki. This same question was asked just a few days before you discovered this mailing list.-Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob HunterSent: Monday, June 14, 2004 4:54 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] making * more like a normal pbxonce u press 9 is there a way to make it so it restores dial tone, like most pbx's do?sodial tone , 9, dialtone, then ur local num--Gafachi.com - referal code hunter81instant iax termination - 2 cents a minuteAlso they have a great referal program,tell them jacob, hunter81 sent you
