Re: [Asterisk-Users] Sometimes calls are silent
Hi Jonathan, can you be a little more clear ? What is your test configuration? How do you expect to have voice if you use only one FXO of the card (maybe you use regular phones too ..) Regards, Ciprian On Wed, 01 Dec 2004 15:51:24 -0500, Jonathan Bartlett [EMAIL PROTECTED] wrote: I'm setting up an asterisk server, used as a gateway to regular phone lines. I've got a TDM400P card with FXO modules, but I'm only using one to test. When I make outgoing calls, occassionally it seems like the incoming audio is switched off. It will work fine for several calls, and then for several more calls the incoming audio will be silent. The other party can still hear me, I just can't hear them. I'm using fxs with ks. I've tried it w/ echo cancelling both on and off. Anyone else have this problem? It never happens in the middle of a call, either. The call either has sound or it doesn't have sound. I'm using Firefly for my SIP phone, but I don't think it's an issue with the soft phone. Thanks! Jon ___ 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] Is anyone using Cisco 7905G phones?
On Mon, 6 Dec 2004 22:45:16 -0800, Randy MacKay [EMAIL PROTECTED] wrote: I have a few Cisco 7905G phones and I having a little trouble configuring them. They are working with Asterisk. I'm able to get the sip image loaded, but I can't get the phones to blind transfer. Does the Cisco 7905G Phone use XML Services? If you are using the 7905G phone, would you post any of your configuration files so I can try and figure out where I'm going wrong? Thanks for your help, Randy I use them, they work fine with blind transfer. They don't, however, use XML services. I'll send an email following this one direct to you with my configuration. -Shaun ___ 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] PBXware
I want to use PBXware but I've found that the version we need is around $1,000. I found quite a few other solutions at http://www.voip-info.org/tiki-print.php?page=Asterisk+GUI . Does anybody have any specific suggestions ? I need a product that's similar or better than PBXware. Best Regards, Alex Brecher http://www.SuccessfulHosting.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] new version problems
Good day all We got the cvs yesterday,and it seems as if transfer does not work.We are using mitel 52205055 and the Grandstream bt-100,using the transfer buttons. If you transfer it just goes to the next step? please Help Thanks Altus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is anyone using Cisco 7905G phones?
Yes I'm working with 7905G phones. There's no problem in transfer calls.
Here's the regarding entry in my sip.conf:
[garage]
type=friend
username=7905g_1
secret=**
host=dynamic
canreinvite=no ; Cisco poops on reinvite sometimes
defaultip=1992.168.1.7
callgroup=1
context=ipphone
But you should'nt forget to enable call transfer within your
dial-statement in extensions.conf:
[ipphone]
exten = 123,1,Dial({$EXTEN}|20|Tt)
Also a problem could be the config on the phone itself. Via the
webinterface you can tell the phone to handle the transfer buttons or
not. Simply go on the page Call Preferences and the last two lines
called GUI Show Mask and mor important GUI Set Mask handle those
settings. You have to enable Bit 10 (Call Transfer) and 11 (Blind
Transfer) and now it will work.
XML services won't work for my 7905G's. Each entry in the config files
is only marked as reserved - may be in future there'll be an usable
entry but till now there're no entries enabled.
Jens
I have a few Cisco 7905G phones and I having a little trouble configuring
them. They are working with Asterisk. I'm able to get the sip image
loaded, but I can't get the phones to blind transfer.
Does the Cisco 7905G Phone use XML Services?
If you are using the 7905G phone, would you post any of your configuration
files so I can try and figure out where I'm going wrong?
Thanks for your help,
Randy
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[Asterisk-Users] Voice mail problem
Hi all of you. I am trying to configure voice mail in asterisk and i am facing problems. I have found following warning message in /var/log/asterisk/messages -- No application 'Voicemail' for extension (macro-mainmenu, s, 5) I have configured voice mail accordingly in extention.conf [headoffice] -- - exten = _63,1,Macro(mainmenu) --- [macro-mainmenu] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,Background(nobodypicked) exten = s,5,Voicemail(s901) in sip.confg [901] context=headoffice type=friend username=901 secret=password host=dynamic qualify=1000 ;new entry [EMAIL PROTECTED] --- in voice mail.conf [headoffice] 901=password(numeric), Mazhar User, [EMAIL PROTECTED],,tz=san-diego|attach=yes and also have made directory /var/spool/asterisk/voicemail/headoffice/901/INBOX/ And also let me know is there any concern of voice mail with sip.conf ? Regards, Mazhar ___ 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] Voice mail problem
No application 'Voicemail' for extension (macro-mainmenu, s, 5) Did you load = app_voicemail.so in your modules.conf? Our simply set autoload=yes? Jens -- Jens Lentföhr http://www.jens-it.de ___ 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] Interface analogue exchange line to VOIP phone?
I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue-VOIP adapter that has an FXO port in it instead of just an FXS port? i.e. designed to connect to an exchange line instead of a phone? The VOIP port on the adaptor would then be made available over the internet, for the remote worker to connect his VOIP phone to. 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] Strange softphone problem
Now here is strange problem i experience. Setup is easy, IAX line out with SIP softphone registered to Asterisk. All work fine except for one client. When using Sjphone the other end can not hear a thing. When using X-pro the opposite happens, local user can not hear a thing. These softphones work fine on other clients on same network. I've also tested several headsets but same outcome. Also same SIP account works on other clients fine. Client is WinXP SP2 with no firewall activated. -- Cinoss [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] Budgetone 100 Caller ID
Hmm, I managed to get callerid working last night!
That is calls coming in from POTS on my X1000P card show up correctly
at ASterisk.
I noticed on by BT102 phone that the number was displayed! Great!
However when I dialled in and withheld my number, the Bt102 showed something
which resembled ' tr1' ?
Is this normal?
thanks
Mike
On Sun, 05 Dec 2004 07:09:53 -0500, Greg - Cirelle Enterprises
[EMAIL PROTECTED] wrote:
At 06:24 PM 12/4/04, you wrote:
Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
Thanks Doug, will try that
Greg
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[Asterisk-Users] gsm codec, very poor quality.
Currently I am creating .wav files and then converting them via SOX to .au file format, then running them througha gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. ___ 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] G.729 algorithm?
Steve Underwood wrote: Albania, I think :-) Cite your source. -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.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
Re: [Asterisk-Users] Interface analogue exchange line to VOIP phone?
I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue-VOIP adapter that has an FXO port in it instead of just an FXS port? i.e. designed to connect to an exchange line instead of a phone? The VOIP port on the adaptor would then be made available over the internet, for the remote worker to connect his VOIP phone to. A pair of Sipura spa-3000's (see forum at voxilla.com for configs). Also, some combo of spa-1000 and spa-3000 is likely to handle it. Lots of other vendors out there doing the same thing. Most of those products have been sold in the past as toll bypass products, but they are doing exactly what you want. ___ 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] chan_misdn and Dynalink IS64PH ISDN
Hi, So if I want NT mode, I need layer 2 and 3 in user space ? Exactly. How can I use the mISDNuser library to works with asterisk ? I have compiled chan_misdn with mISDNuser. That should be enough. A nother question, to connect asterisk to a classic pbx, what I need ? NT or TE mode ?? ptp mode ? both (NT + ptp ?) You want TE (Terminal Equipment) mode if the PBX sees Asterisk as a phone and NT (Network Termination) mode if the PBX sees Asterisk as the ISDN network. In both cases, PMP (Point-to-MultiPoint) mode should be used. Simon signature.asc Description: OpenPGP digital signature ___ 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] High(er) availability
Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -- Asterisk SIP carrier | | --- Asterisk (standby) -- Asterisk is used as a PABX in this setup, so the sip phones register themselves at the asterisk machine and the asterisk machine calls out if necessary. What I would like to be able to do is if the first asterisk machine fails I want to have a 2nd machine standby. So the standby asterisk monitors the first asterisk and in case of a failure the standby asterisk takes over the IP of the 1st asterisk so the services continues (sync the conf file with rsync for example), however if the phones use a host=dynamic they wont be able to be called until they have reregistered themselves at the backup asterisk. Is there a SER like t_relay kinda thingy to let the backup know the locations of the Sip Phones? Kind regards, E. Versaevel ___ 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] Re: dont write me again
On Wednesday 01 December 2004 19:44, Stephen R. Besch wrote: Exactly. Would those people who respond from the mailing list digest -PLEASE-PLEASE-PLEASE- do the following simple things: 1)Strip out the digest messages that have nothing to do with your reply. 2)Copy the appropriate subject line into your message subject before you send the message so that we can actually tell what you are writing about without having to search through your reply. you missed out the most simple read the damn email. It clearly tells you howe to unsubscribe Jon ___ 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] Interrupt latency problems
On Wednesday 01 December 2004 20:31, Steven Critchfield wrote: I am glad it solved the problem. Now if only someone knew what it was about the stock RH or FC kernel that makes it happen you could get RH or FC to stop using that patch. That or maybe more people will be like me and always cast a weary eye upon a prepackaged kernel no matter what distro it came from. First thing when installing any distro is to bin the kernel and install a vanilla one - how else can you be sure of the state of possibly the most important part of your system. Jon ___ 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] GrandStream BT VS. IP500 Latency
I just noticed something when I 'sip show peers' from the CLI, I get the following: 6113/6113 x.x.x.x D N 255.255.255.255 62927OK (66 ms) 6112/6112 x.x.x.x D N 255.255.255.255 50079OK (160 ms) 6111/6111 x.x.x.x D N 255.255.255.255 60810OK (141 ms) 6109/6109 x.x.x.x D N 255.255.255.255 51331OK (151 ms) All of those are behind the same firewall (openbsd 3.6, pf, nat) and on the same network with no QOS (yet). 6113 is the Grandstream BT. I am just wondering why the times are significantly lower on the ping to it than any of the other phones. All of the other phones are IP500's with the newest public firmware release. Thanks, Matt -- Matt Gibson VOIP Administrator NJ Tech Solutions 1.314.480.4550 ex. 6400 1.877.999.4678 ex. 6400 ___ 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 and Cisco IP Phones
On Tue, 7 Dec 2004, Julien Goodwin wrote: On Mon, Dec 06, 2004 at 07:43:24AM -0600, Rich Adamson arranged a set of bits into the following: I don't think its an argument as much as it is folks expressing opinions without giving you a clue why they've formed that opinion. Here's another one. SCCP is a cisco proprietary protocol that some folks have partially reverse engineered, writing * code to support those basic functions that have been reverse engineered. Not all of Cisco's SCCP functions have been reverse engineered. If you compare functionality of what Actually one you get past registration of the phone (which can sometimes be pretty odd) we can support almost everything. The problem is that we don't have the function code written yet to support call forwarding or ad-hoc contrences (and similar), this generally doesn't take very long to write but has to be solidly tested due to all the protocol differences between the phones. Is there any way to make chan_sccp log all of the communication between chan_sccp and the phone? I did not manage to get my Kirk IP600 to register when using chan_sccp (with built-in skinny it does register) and would like to try to get it to work (my evaluation period on the unit is expiring soon). 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
[Asterisk-Users] Mini-ITX Mainboard for Asterisk IP PBX, Intel Mobile Celeron 733MHz
Hi, I would like to offer you the following specialized embedded Mini-ITX Mainboard: Samples: $390 50 pcs: $270 100 pcs: $255 The Technical Specification is: Dimension: Mini-ITX, 170x170mm System Processor:Intel Mobile Celeron 733MHz (Fanless) Chipset: Intel 830M + ICH4 BIOS:Award Flash 256K BIOS System Memory: One DIMM socket for SDRAM memory module up to 512MB Display Controller: Intel 830M integrated graphics contoller CRT: Integrated 350-MHz RAMDAC, supports progressive scan analog monitor up to a resolution of 1800 x 1440 pixels LCD: Onboard LVDS Transmitter through DVO port TV: Onboard TV-out encoder Focus FS454 through DVO port Ethernet Controller: Two PCI-bus Ethernet controllers realtek RTL8100C, one for WAN with Power over Ethernet (IEEE 802.3af) and the other one is for LAN Sound Output:AC 97 V2.3 for Line-out, line-in and Mic-in CompactFlash:One type-II compact flash socket IDE Interface: 2 x IDE ports Serial-ATA: 1 x Serial ATA port USB: 6 x USB 2.0 IR Interface:1 x IrDA Expansion slots: 1 x PCI slot for PCI Raiser Card with 3 PCI 1 x Mini-PCI Power Connector: DC-in Jack for DC +48V Power management:ACPI function RTC: LPC Super I/O including Hardware Monitor:LPC Super I/O including Operating Temp.: 0-60 degree C Humidity:5-95% RH, non-condensing It is possible to be added 3rd Ethernet port for DMZ or other purposes. Best Regards, Miroslav Nachev ___ 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] Budgetone 100 Caller ID
Thats normal when it cant discover the ID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Dent
Sent: Tuesday, December 07, 2004 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Budgetone 100 Caller ID
Hmm, I managed to get callerid working last night!
That is calls coming in from POTS on my X1000P card show up correctly
at ASterisk.
I noticed on by BT102 phone that the number was displayed! Great!
However when I dialled in and withheld my number, the Bt102 showed something
which resembled ' tr1' ?
Is this normal?
thanks
Mike
On Sun, 05 Dec 2004 07:09:53 -0500, Greg - Cirelle Enterprises
[EMAIL PROTECTED] wrote:
At 06:24 PM 12/4/04, you wrote:
Greg - Cirelle Enterprises wrote:
Hi,
Is there an * configuration that will allow the BT100 to
display the numeric callerid instead of the broken
text?
exten = extension,priority,SetCIDNum(${EXTEN})
Doug
Thanks Doug, will try that
Greg
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[Asterisk-Users] Asterisk 1.0.3
Greetings! Version 1.0.3 has been released of Asterisk, Zaptel, and libpri. As usual, the tarballs can be downloaded from the Digium ftp server. For more detailed download instructions, please see http://www.asterisk.org/index.php?menu=download. The changes to Zaptel and libpri are minor. However, there are a significant number of bug fixes for Asterisk in this release. The changes for all three are summarized in the ChangeLog, which is on the ftp server as well as included in the tarball. There is a new channel on irc.freenode.net for issues directly related the the stable branch of Asterisk. It is #asterisk-stable. I hope everyone finishes out their year on a happy note. Mark Spencer and I had some fun with Christmas decorations the other night. He now has a huge, inflatable penguin display in his front yard accompanied by an Asterisk made out of Christmas lights. Both of them are about 12-feet wide. See the following link for a fuzzy picture from a cell phone that was taken in the rain: http://www.marko.net/~mark/markyard.jpg Cheers! Russell Bryant drumkilla ___ 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 0.3.5 does not compile
I use stable CVS asterisk and it is working without problems. But now i am trying to compile chan_capi 0.3.5 module and i get following error /usr/src/chan_capi-0.3.5# makegcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.cchan_capi.c:23: asterisk/features.h: No such file or directorychan_capi.c:24: asterisk/utils.h: No such file or directorychan_capi.c: In function `restart_monitor':chan_capi.c:2278: warning: implicit declaration of function `ast_pthread_create'make: *** [chan_capi.o] Error 1 I am using debian stable, kernel 2.4.28 greetings Milos ___ 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] PRI/Zap premature dialing problem
Peter, thanks for educating this ISDN-ignorant American! The ASCOM and the problem are in Germany. This is definitely overlap dialing in the extreme, from looking at the PRI debug output of asterisk. I set overlapdial=yes in zapata.conf, with no difference observed in the behavior. Here's the PRI debug output. Each dialed digit (from an Ascom desk phone) is transmitted separately, immediately when the user presses the key. (essentially the same as if it were sending DTMF). Asterisk is jumping instantly on the first match, even tthough there is a pattern _2XXX in the same dialplan, with higher priority (the 224 is in an include= context to suppress it). So when I want to get to extension 2246, I have no chance, it immediately jumps to 224. I agree that modifying the ASCOM behavior would probably be better, but that is a slow and expensive strategy, there must be some way to get Asterisk to play well with this simple digit-by-digit inbound dialing method... - PRI DEBUG LOG...Please excuse the verbosity!! - Protocol Discriminator: Q.931 (8) len=30 Call Ref: len= 2 (reference 36/0x24) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] Calling Number (len=11) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Unknown (52) '1520440' ] IE: High-layer Compatibility (len = 4) -- Making new call for cr 36 -- Processing Q.931 Call Setup -- Processing IE 4 (Bearer Capability) -- Processing IE 24 (Channel Identification) -- Processing IE 108 (Calling Party Number) -- Processing IE 125 (High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32804/0x8024) (Terminator) Message type: SETUP ACKNOWLEDGE (13) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 36/0x24) (Originator) Message type: INFORMATION (123) Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2' ] -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 36/0x24) (Originator) Message type: INFORMATION (123) Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2' ] -- Processing IE 112 (Called Party Number) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 36/0x24) (Originator) Message type: INFORMATION (123) Called Number (len= 4) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4' ] -- Processing IE 112 (Called Party Number) -- Executing Goto(Zap/31-1, default|nikoflat|1) in new stack -- Goto (default,nikoflat,1) -- Executing Dial(Zap/31-1, IAX2/pbx:[EMAIL PROTECTED]/nikoflat|16) in new stack -- Called pbx:[EMAIL PROTECTED]/nikoflat -- Accepting call from '41520440' to '224' on channel 31, span 1 -- Call accepted by 192.168.50.254 (format ULAW) -- Format for call is ULAW -- IAX2[192.168.50.254:4569]/1 is ringing Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32804/0x8024) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] Protocol Discriminator: Q.931 (8) len=14 Call Ref: len= 2 (reference 32804/0x8024) (Terminator) Message type: ALERTING (1) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 31 ] Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate
Re: [Asterisk-Users] gsm codec, very poor quality.
Sorry this doesn't answer your question. Any reason to not leave them as wav's? On Tue, 7 Dec 2004 10:42:58 +0100, Matthew Oulton [EMAIL PROTECTED] wrote: Currently I am creating .wav files and then converting them via SOX to .au file format, then running them through a gsm codec convertor which all works fine except that it sounds like the recording was made with a sock in my mouth !! Could someone in * land help me to get a good sound quality with gsm format. Thanks in advance. ___ 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 -- Is it something someone said, was it something someone said? ___ 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] Kind of off-topic: VoIP services and multiple callers
On Monday 06 December 2004 22:59, Rich Adamson wrote: Inline... I know that VoIP providers can supply their customers with a local number and/or virtual numbers, and then that number/account can be used with Asterisk (well, it depends on the provider and whether or not their service is compatible with Asterisk). However, I have a question: can more than one person make/receive a call at the same using one VoIP line? Some providers support multiple calls, others don't. If five people in the office all need to use their phones at the same time, would I need five VoIP lines, or would I only need one VoIP line? Am I over-thinking this? Same answer. Its up to the provider, and when they do support multiple calls, they typically charge a fee/minute/call so its no skin off their back. voiptalk.org allow 2 calls over a standard (free) account. You can get more calls allowed for a fee. I'm sure there are others that do similar. Jon ___ 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: Interface analogue exchange line to VOIP phone?
In article [EMAIL PROTECTED], Rich Adamson [EMAIL PROTECTED] wrote: I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue-VOIP adapter that has an FXO port in it instead of just an FXS port? i.e. designed to connect to an exchange line instead of a phone? The VOIP port on the adaptor would then be made available over the internet, for the remote worker to connect his VOIP phone to. A pair of Sipura spa-3000's (see forum at voxilla.com for configs). Also, some combo of spa-1000 and spa-3000 is likely to handle it. Thanks Rich, looking at the Sipura site is looke like the SPA-3000 should do what I'm looking for, with its FXO port connected to an extension line on the analogue PBX, and its Ethernet port exposed to the Internet (with suitable security) for the remote worker to point his VOIP phone to. Lots of other vendors out there doing the same thing. Most of those products have been sold in the past as toll bypass products, but they are doing exactly what you want. I understood the toll bypass products as being for connecting an analogue *phone* to a VOIP network (e.g. ATA-286), which was the opposite of what I wanted. Anyway, the SPA-3000 looks ideal - thanks! 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] callerid PSTN-IAX problem
Hi, I cannot see cid for incomming call from PSTN (Quintum gateway) to IAX client (FireFly). Client displays blank but when I look into cdr's /var/log/asterisk/cdr-cvs/Master.cvs, the callerid is registered properly. Why it's not displaying? L. ___ 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] IAX2 Hangup Cause
Dear List im try to look if IAX2 protocol is able to transport an hangup cause from a TDM PRI line, as i can see from this link http://www.cornfed.com/iax.pdf seems support only few message like congestion,busy,call progress, answer,ring,ringing but i cannot transport the Cuase of PRI hangup some idea. IF i use the send text feature of IAX2 between 2 * Box? I can receive this text? I design my scenario: Site A (AstGateway) Site B (AstGateway) Incoming-TDM- IAX2 |---|Hangup Cause|--| IAX2-TDM - Termination Thanks for possible help Dimitri ___ 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] two questions
On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ 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] Transfer on Snom 190
I cannot get the transfer button to work on a Snom 190, I cannot get the # to work either. Any ideas? Regards Thorben ___ 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: Asterisk and Cisco IP Phones
Julien Goodwin [EMAIL PROTECTED] writes: Otherwise you can let us know what's missing for you and we'll see what we can do. Since you ask... :-) I'm using chan_sccp with an old 12SP+, and it's working fine except that no ring or busy signal is heard when dialing out from the phone. On console: -- Called [EMAIL PROTECTED] -- SIP/voop-gw-01e1 is making progress passing it to Skinny/[EMAIL PROTECTED] -- Asked to indicate 'UNKNOWN' condition on channel Skinny/[EMAIL PROTECTED] Not surprisingly, nothing is heard until the call is answered. -tih -- Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting www.eunet.no T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145 ___ 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] Comdial PBX -- can use Asterisk as VM box?
Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a KeyVoice application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the hardware. Is there any way to get Asterisk to respond to whatever mechanism it is that the Comdial puts out to the Dialogic? Things I've already tried and discarded: DID: the PBX strips off the DID stuff before it gets to the Asterisk box Caller ID: ibid. So, I'm guessing that there's some, for lack of a better word, protocol that must be standardized to some extent, that allows things like the Comdial PBX to talk to someone else's VM solution. Can Asterisk play ball? Thanks! -Ken ___ 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] Question about e1/digium
Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to direct for a telephone sip ? Thanks for attencion Sergio Faulhaber [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] Re: Interface analogue exchange line to VOIP phone?
I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue-VOIP adapter that has an FXO port in it instead of just an FXS port? i.e. designed to connect to an exchange line instead of a phone? The VOIP port on the adaptor would then be made available over the internet, for the remote worker to connect his VOIP phone to. A pair of Sipura spa-3000's (see forum at voxilla.com for configs). Also, some combo of spa-1000 and spa-3000 is likely to handle it. I suggest you log into the voxilla.com site and read the postings from folks that have already done this to ensure you purchase the right boxes. I've not tried this personally. Thanks Rich, looking at the Sipura site is looke like the SPA-3000 should do what I'm looking for, with its FXO port connected to an extension line on the analogue PBX, and its Ethernet port exposed to the Internet (with suitable security) for the remote worker to point his VOIP phone to. Lots of other vendors out there doing the same thing. Most of those products have been sold in the past as toll bypass products, but they are doing exactly what you want. I understood the toll bypass products as being for connecting an analogue *phone* to a VOIP network (e.g. ATA-286), which was the opposite of what I wanted. Anyway, the SPA-3000 looks ideal - thanks! The toll bypass products typically do not show up on this list since their target audience is certainly not asterisk users. The Mediatrix 1104 and 1204 are examples of such four-port boxes, but there are lots of others. They typically don't use words like voip in their marketing materials and are generally limited to an rj11 here and another rj11 over there, passing some sort of voice encoded packets over IP. They are oftentimes sold by traditional pbx resellers. For your objective, the sipura products are about as cheap as you can get. (Or, should I say 'inexpensive'; not sure. :) ___ 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] Linking asterisk to an existing small office PBX
Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have several analogue extensions spare, but no capacity to add fancier connectors to link to an asterisk system (as most of the PBX linking guides detail). All our phones are bog standard analogue ones. We'd like to use an asterisk system to allow some calls to be routed out via a VoIP gateway. We'd also like to allow some inbound SIP calls to be handed to the PBX. I was considering building an asterisk system, and adding a few FXS cards to it. I'd plug these into the spare analogue extensions on the PBX (is it FXS, and not FXO I'd need when connecting to a PBX?) If people wanted to make a call using a VoIP gateway, they'd dial one of the asterisk extensions. They would then be connected to the asterisk box by the PBX, and could dial the real number they wanted. Finally, asterisk would connect them to an external SIP gateway, which would do something useful with the call. Additionally, I'd like people working from home to be able to connect via SIP to the asterisk box, and then have their calls routed to the PBX down an analogue line. Is this possible, and is it even a desirable setup? Thanks Nick ___ 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] G729, x-pro, and codec ordering
Try setting the codec settings for each peer instead of under the general heading. On Tuesday 07 December 2004 05:39 am, Paul Fielding wrote: I'm in the middle of getting g729 to work on my server and running into odd stuff. The issue revolves around what appears to be a much talked about (but not seeming to be much solved) issue of selecting which codec gets used at a given time. I have two g729 licenses. I'd like to be able to get asterisk to use g729 (via x-pro) only when I want to, reason being that if I'm in a high bandwidth environment I'd rather have the higher quality of ulaw, but when I'm in a low bandwidth environment I'd like to select g729. There doesn't seem to be much rhyme or reason to which codec gets chosen, and it seems to vary depending on whether the call is outgoing or incoming. And furthermore, turning off a codec in x-pro doesn't seem to do anything. For example, if I have: [general] disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm allow=ilbc and then dial out on x-pro, G729 is selected. Then I turn off G729 and turn on g711u (I make g711u the only black codec on the x-pro display), then make a call, the call is still made using G729. Further more, with the same settings if I call from a zap channel to the x-pro sip extension, the codec chosen is g711u, even though I might only have g729 enabled on x-pro, and even though g729 is the first one on the list above. Anyone have any suggestions, or can point me to something to read? regards, Paul ___ 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 -- Brian Wilkins Software Engineer [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 321.308.4000 x33 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi 0.3.5 does not compile
On Tue, 2004-12-07 at 11:36 +0100, Milos Kocbek wrote: [snip] chan_capi.c:23: asterisk/features.h: No such file or directory chan_capi.c:24: asterisk/utils.h: No such file or directory [snip] Iirc you don't have the asterisk header files installed. They are installed when you do make install in the asterisk src directory. Install them and try again. Regards, Patrick ___ 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] Linking asterisk to an existing small office PBX
I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have several analogue extensions spare, but no capacity to add fancier connectors to link to an asterisk system (as most of the PBX linking guides detail). All our phones are bog standard analogue ones. We'd like to use an asterisk system to allow some calls to be routed out via a VoIP gateway. We'd also like to allow some inbound SIP calls to be handed to the PBX. I was considering building an asterisk system, and adding a few FXS cards to it. I'd plug these into the spare analogue extensions on the PBX (is it FXS, and not FXO I'd need when connecting to a PBX?) If people wanted to make a call using a VoIP gateway, they'd dial one of the asterisk extensions. They would then be connected to the asterisk box by the PBX, and could dial the real number they wanted. Finally, asterisk would connect them to an external SIP gateway, which would do something useful with the call. Additionally, I'd like people working from home to be able to connect via SIP to the asterisk box, and then have their calls routed to the PBX down an analogue line. Is this possible, and is it even a desirable setup? Yes, all of that is possible with lots of folks already doing it. The extension appearances on your existing pbx are fxs, therefore the mating interface on asterisk has to be an fxo. The digium TDM04B is one example of a 4-port pci card supporting 4 fxo interfaces. If each of those four fxo interfaces were connected to unused extensions from your old pbx, incoming voip calls (via the Internet) can be routed to any of those four fxo ports. Likewise, an existing pbx user could dial 8 as an example, and be sent to asterisk via those same four ports. You would need to be able to set up the dialplan in both the old pbx and asterisk to handle the exact dialed digits the way that you want. It is highly unlikely the old pbx can be made to forward callerid numbers to asterisk, etc. As time and budget permit, you could migrate to using SIP phones on the asterisk side as well, displacing the old pbx analog phones. ___ 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] Linking asterisk to an existing small office PBX
On Tue, 7 Dec 2004, Nick Burch wrote: Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have several analogue extensions spare, but no capacity to add fancier connectors to link to an asterisk system (as most of the PBX linking guides detail). All our phones are bog standard analogue ones. We'd like to use an asterisk system to allow some calls to be routed out via a VoIP gateway. We'd also like to allow some inbound SIP calls to be handed to the PBX. You could put Asterisk between the old pbx and the incoming isdn lines. A four port isdn card would be used in the Asterisk box. This way all functions should be available to both the old pbx and the new Asterisk box. 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
[Asterisk-Users] Another Unable to create channel of type 'Zap' (cause 0) error
.. and from a newbie no less :-) I have configured my BT101, and hooked it up to my * box. All is well. I have entered the following in externsions.conf, and this bit works: exten = 613,1,Answer exten = 613,2,Playback(demo-echotest) exten = 613,3,Echo exten = 613,4,Hangup If I pick up the BT101, and dial 613, sure enough I get the echo test.. All good. I have a TDM400 Card with a single FXO port on it. ztcfg -vv recognises the card as FXS Device (I think that's right though...??) I want to know how to get, say extension 1000 to dial a number on the FXO card.. ie: exten = 1000,1,Answer exten = 1000,2,Dial(Zap/1:555-1234,20,tr) exten = 1000,3,Hangup That should work, shouldn't it? Well it doesn't :-) Hence the error in the subject of this message!.. I'm a total noob, but once I get my head around this, I'm sure I'll have no problems.. Oh, and what extension do I use to reference an incoming call on my FXO port? exten = 1 ?? Alan ___ 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] Polycom IP500
http://www.freedomphones.net/polycom/files/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Sent: Sunday, December 05, 2004 4:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Polycom IP500 Does anyone have a location to download the latest Polycom firmware etc? Other than the extranet site, because I am not a reseller, there fore I have no login. [minirant] And shouldn't end users be granted access to this kind of thing anyway? Geeze [/minirant] Thanks, Chris Cherry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.289 / Virus Database: 265.4.5 - Release Date: 12/3/2004 ___ 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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] Linking asterisk to an existing small office PBX
On December 7, 2004 07:51 am, Nick Burch wrote: Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have several analogue extensions spare, but no capacity to add fancier connectors to link to an asterisk system (as most of the PBX linking guides detail). All our phones are bog standard analogue ones. Ok so you have two ISDN PRI or two ISDN BRI? (i.e. how many simultaneous calls can you make or take to the phone network?) And you're saying you have 24 regular old phones -- the kind you can plug in to a regular phone jack and use normally. There are no soft buttons or fancy displays or anything? We'd like to use an asterisk system to allow some calls to be routed out via a VoIP gateway. We'd also like to allow some inbound SIP calls to be handed to the PBX. No problem. Although I would not use FXS cards -- with 24 phones that is ripe for a channel bank, and since they're FXS and not FXS pretty much any channel bank you can find will work just fine; I recommend the Adit600 personally but they are pricier than the older Access Bank I and II (I handles 1 T1, II handles 2) -- ABI/IIs can handle FXS lines without any issues whatsoever. They don't work worth a shit for FXO ports though, since they don't have functioning far-end disconnect supervision (i.e. they can't tell when the other side has hung up). So a T100P and an ABI will handle all your existing phones without any worry whatsoever. Price: US$500 for the T100P and ~US$250 or so for an ABI off of ebay. Any old Asterisk box will handle SIP phones, so as long as you have an ethernet card it'll work. Depending on what you have for incoming lines (see my question above) you'd either use a T100P (total 3, may as well get a TE405P) or a single Sangoma A102u (2 T1s in 1 PCI card), or some kind of ISDN BRI card -- I am *not* familliar with the ISDN BRI stuff, so I'll defer that to someone else. Depending on what your existing KSU or PBX is doing you can get rid of the thing altogether and let Asterisk do all your phone stuff, or try and integrate the two. I have successfully integrated * with Norstar MICS (PRI and POTS) and am currently working on an NEC system whose model name escapes me at the moment. The only reason I'm integrating them instead of replacing them is that the people I'm doing this for are quite fond of their digital phones. :-) -A. ___ 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] G.729 algorithm?
Kevin Walsh wrote: Robert Rozman [EMAIL PROTECTED] lazily top-posted: do you have info in what countries g.729 is not valid... ? You could start with the whole of Europe and can also add the UK. I'm sure there are lots of other countries who don't feel the need to acknowledge US-based software and algorithm patents too. This subject has been covered several zillion times in the mail list. Google is your friend. Would that be the same UK as the one that came up with this: http://www.*patent*.gov.*uk*/*patent*/ legal/decisions/2004/o29204.pdf ? The worrying thing about that is within the arbitrators terms of reference the decision is right. This is a patentable thing under the 1977 UK patent law. However, prior art goes back to the earliest computers in the 1940s. The decision doesn't seem to allow for that. Regards, Steve ___ 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] High(er) availability
On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote: Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? Basically I would start with building redundancy in the the primary server, e.g. a ton of fans so one can break down without frying the box, redundant hot swappable power supplies, hot swappable disks in RAID1 or something like that. That will reduce the chance of the primary server going down due to hardware problems. Interesting problem as you need to be able to preserve state across multiple servers. Did you look at that realtime app that's part of CVS (maybe asterisk-addons)? If it stores the registration state of the phones in a DB then both servers should have no problem being aware of all regs if one of them fails. Question is how do you make the backup server's Asterisk listen on the newly assigned IP of the primary server? Prolly a sip reload would solve that. Wouldn't SER be a better option? Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Regards, Patrick ___ 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] G.729 algorithm?
Eric Wieling aka ManxPower wrote: Steve Underwood wrote: Albania, I think :-) Cite your source. I might be wrong. I'm working from second hand knowledge. Someone told be they never introduce copyright legislation and their patent legislation is almost non-existant. I think you would be in the clear there, but before proceeding I suggest you consult an Albanian lawyer. :-) Stve ___ 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] Another Unable to create channel of type 'Zap' (cause 0) error
On Mon, 2004-12-06 at 17:40, Alan Ingleby wrote: exten = 1000,2,Dial(Zap/1:555-1234,20,tr) Change this to exten = 1000,2,Dial(Zap/1/5551234,20,tr) Oh, and what extension do I use to reference an incoming call on my FXO port? exten = 1 ?? You want the s extension. http://www.voip-info.org/wiki-Asterisk+s+extension -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Skinny error : Unable to create channel
Hi list! I'm getting these errors in the log: Dec 7 11:08:04 NOTICE[442388]: No available lines on: [EMAIL PROTECTED] Dec 7 11:08:04 NOTICE[442388]: Unable to create channel of type 'Skinny' What does this mean? Cheers! Remco ___ 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] H.323 trunking
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx should be
routed to the H.323 trunk.
thx for your feedback
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RE: [Asterisk-Users] High(er) availability
On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote: Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? Basically I would start with building redundancy in the the primary server, e.g. a ton of fans so one can break down without frying the box, redundant hot swappable power supplies, hot swappable disks in RAID1 or something like that. That will reduce the chance of the primary server going down due to hardware problems. That would be logical, also at least two ups (1 for each powersupply) Interesting problem as you need to be able to preserve state across multiple servers. Did you look at that realtime app that's part of CVS (maybe asterisk-addons)? If it stores the registration state of the phones in a DB then both servers should have no problem being aware of all regs if one of them fails. Haven't looked into that, I believe that's for realtime reading of the config files (which isn't realy an issue, just rsync em) Question is how do you make the backup server's Asterisk listen on the newly assigned IP of the primary server? Prolly a sip reload would solve that. Wouldn't SER be a better option? I'm thinking of giving the backup ser an alias the same as the primary, but it should not respond to ARP request (so no packets get there), in case of a failure it should start responding to ARP. SER isn't an option, I need a PABX, not a SIP Proxy. Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. Regards, Patrick Regards, E. Versavel ___ 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] Website that reads text recently on the list?
there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. ___ 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] High(er) availability
E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Stefan de Konink ___ 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] save dialplan missing in 1.0.2??
I seem to be missin the save dialplan command in asterisk 1.0.2, I have been searching for info but all I get is how to use it. Anybody have any info on this? Regards Greg Cirino ___ ___ 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] Website that reads text recently on the list?
On Tue, Dec 07, 2004 at 09:44:59AM -0500, Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. I think it was an ATT research site. Steve -- NetTek Ltd Phone/Fax +44-(0)20 7483 2455 SMS steve-epage (at) gbnet.net [body] gpg 1024D/468952DB 2001-09-19 ___ 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] H.323 trunking
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] Website that reads text recently on the list?
http://www.rhetorical.com/cgi-bin/demo.cgi Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. ___ 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] High(er) availability
That would lead more to keepalived I think Would be an option, but I would have to use fixed IP addresses for the IP Phones (that should not be a problem) Erik E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Stefan de Konink ___ 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] Website that reads text recently on the list?
Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. http://www.babeltech.com/Demos.php?s=48m=3f=95 http://www.scansoft.com/realspeak/demo/ Stefan de Konink ___ 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] High(er) availability
On Tuesday 07 December 2004 14:39, E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. Take a look at keepalived I've used it (along with it's implementation of VRRP) to provide failover for routers. I see no reason why it couldn't do the same for an asterisk server. You might have to write a module to monitor the actual asterisk process. Jon ___ 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] modprobe ztdummy - failed
Title: Message Hi all, I have a problem starting the ztdummy. Here is what happens: [EMAIL PROTECTED] /]# modprobe ztdummyNotice: Configuration file is /etc/zaptel.confline 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing '/etc/asterisk/meetme.conf': FoundDec 7 15:44:01 WARNING[18359]: chan_zap.c:775 zt_open: Unable to open '/dev/zap/pseudo': No such file or directoryDec 7 15:44:01 ERROR[18359]: chan_zap.c:6811 chandup: Unable to dup channel: No such file or directoryDec 7 15:44:01 WARNING[18359]: app_meetme.c:229 build_conf: Unable to open pseudo channel - trying deviceDec 7 15:44:01 WARNING[18359]: app_meetme.c:232 build_conf: Unable to open pseudo device I have used following command to make the ztdummy: make clean make linux26 make install I use Fedora Core 3. Regards, Stojan Sljivic ___ 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] Website that reads text recently on the list?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: Wednesday, 8 December 2004 1:48 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Website that reads text recently on the list? On Tue, Dec 07, 2004 at 09:44:59AM -0500, Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. I think it was an ATT research site. Steve -- The ATT link is.. http://www.research.att.com/projects/tts/demo.html Cheers Shane ___ 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] Door buzzer.
Henry Devito schrieb: On Sat, 4 Dec 2004, Cian O'Sullivan wrote: They have a pizza box server as their asterisk server with a T1 card. No more slots, so if I want to use the existing infrastructure I will need to build a second server with an FXO port. Kinda stupid having a second server just to open the door. If the device is only a buzzer, can't you do anything fancy on the comport, with hardware and an event poll? Or if it is a phone device maybe an Iaxy can do the trick? Stefan de Konink [*] Viking electronics makes a device for this purpose. www.vikingelectronics.com They make 2 devices, one that will work hooked to an FXS port the other will work hooked to an FXO port. I prefer the one that hooks to an FXO port, But the FXS port box works just as well. Could you expand on this a little? For us that are not into this type of stuff. What exactly is the device one would need, and how could it be installed? -- Best Regards, Mit freundlichen Grüßen, Timm Gebhart [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] TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
Asterisk and it works fine untill the following situation: - one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that line) - both via old and new PBX. - zap show channel n would show that line as 'Offhook', though no telephone is off hook. If physical line would be unplugged from TDM card, the line would become normal again. The offhook in Zap Show Channels is only for FXS cards Sorry, if it is a well known problem, but I did not find any specific information yet.. Please answer two questions: - is it really bad to have parallel connection on TDM400P FXO lines to an additional telephone equipment, does it prevent TDM400P to detect Offhook/Onhook correctly? I have equipment in parallel with no problems, which version of Asterisk are you running Which country are you in? Are you using CallerID facilities if so what signalling/method are you using? - will the problems go away when parallel lines would be disconnected (legacy PBX shut down)? As you may understand the office personnel has anxiety that this may be a bad Asterisk setup / bad TDM card etc (which I am sure so far that it is not). Maybe but as I say I have not seen such problems. Did lines mute before Asterisk? Regards Ian ___ 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] Firewall traversal anomalies - AJA
I'm trying to setup a Cisco ATA 186 which has a public IP address but sits behind a firewall and connects to an Asterisk server with a NAT IP address sitting behind a BSD firewall. The Cisco registers with the Asterisk server without any problems, and I can place calls without any problems and the phone on the other end rings correctly. However, I cannot hear anything through the Cisco after the connection is made. Where should I begin looking for the problem? This is the sip.conf entry for the Cisco: [6184341501] callerid=GlobalEyes 6184341501 canreinvite=no context=from-internal dtmfmode=rfc2833 host=dynamic mailbox=x nat=yes port=5060 secret=xxx type=friend username=x allow=all ___ 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] High(er) availability
This is a good question that the OP posted. Let say you have installed an Asterisk box at a customer location because they have 50 extensions and all talk to eachother alot. If their asterisk box fails, how can you re-direct them to your master box downtown? Matthew - Original Message - From: Stefan de Konink [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, December 07, 2004 8:47 AM Subject: Re: [Asterisk-Users] High(er) availability E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Stefan de Konink ___ 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] zaptel and low ring voltage
Title: OpSign I'd plug four telephones in these lines and test if the lines are really engaged or not and in case it is busy, the other will ring or it will bring you to the voicemail. I ha a similiar problem, the telco had no engaged the lines properly, after this was solved , I also had a damaged FXO channel. Can't you replace the card and see what happens? The telco could also have sold more lines that the switch really supports, thus causing sometimes this problem. I have seen this happening with my local telcos. []s. Jim Van Meggelen wrote: [EMAIL PROTECTED] wrote: Hi all, Several months ago we built an * box with a quad-FXO tdm400p (REV e/f). From the get-go, there has been a problem where occasionally (2-3 times a week) zaptel/* will not detect the ringing on a line. (The call will ring through to telco voicemail). The problem is not specific to a single line or FXO port on the tdm400p. I have 2 theories: #1 - the ring voltage for some calls is below acceptable levels Possible, but also possible that there is too much loss on the circuit. You can test the ringing voltage with a meter, it needs to be between 90V and 110V. Beyond that you may need to use a transmission test set (such as a Wilcom T136B). I got mine for $20 bucks on eBay. Using a butt set and the test set you'll need to call a 1004Hz source from TELUS and then check that you're within the following specs: Loop mA: 23 or better (too hot is no good either, but I doubt that's your problem) Circuit loss: between 0 and -8dB. 0 is really too hot, -3 to -6 is nominal, -8.5 is pushing it, but still within spec. #2 - the tdm400p card is bad Assuming #1, can the zaptel driver be tweaked to be more sensitive to ringing? Any other ideas or experiences? Running asterisk/zaptel v1.0.2 Thank you, 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 -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[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] modprobe ztdummy - failed
Stojan Sljivic - Pamet wrote: Hi all, I have a problem starting the ztdummy. Here is what happens: I have used following command to make the ztdummy: make clean make linux26 make install I use Fedora Core 3. You need to read the udev.README file in the zaptel make directory. Doug ___ 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] High(er) availability
On Tue, 2004-12-07 at 15:47 +0100, Stefan de Konink wrote: E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Cisco claims that VRRP falls under one of their patents, so it could become an expensive option. There are several options out there at this point though that may be able to handle the needs for pre-empting the IP address. About 1 year ago the OpenBSD project wrote a patent-free alternative for VRRP called CARP. It allows for sharing of and automatic failover on an IP address. I have used it to build redundant firewalls that don't lose any state information when the connection drops. CARP is of course built into OpenBSD however I did find what looks to be a userland implementation for Linux. See www.ucarp.org for more information. There are other possible solutions as well, unfortunately I have not used any of these solutions they are just from brief google search. LVS (Linux Virtual Server) mentions VoIP services however I do not know if Asterisk would run in a cluster environment. There are also several sites that deal with high availibity from linux, the first one I noticed that looked like it had some really valuable information is www.linux-ha.org. Unfortunately this is all the easy part. The difficult part will be getting Asterisk to handle the failover gracefully. You probably don't want to lose all the SIP registration data and I have no idea if it will be possible to prevent you from losing the calls. You haven't named that as one of your goals, but it is always something to think about. -- Tim Donahue [EMAIL PROTECTED] Haynes Group, Incorporated ___ 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] Are there any digital phones that run on asterisk yet?
Asterisk can work with ADSI phones, What I have in mind is a pci card with zap-like-driver that supports digital phones. This eliminates (is compairable to using channel bank) additional delay and a primary echo source when both haves of a conversation are carried on the same pair as found with fxs ports. There are many reasons why this is superior to the channel bank/analog phone route especially when VoIP comes into the picture. This is also superior to an office filled with IP phones and a centralized asterisk switch - although there are times where it is clearly advantageous to use an IP phone, such as distant office phones or if all of your calls are destine to IP. But if you are connecting to a telco, calls that avoid IP are always going to be superior and more reliable (with the current state of things). Anyway, there are many pci/digital phone key system or pbx manufacturers. At some point they will see the inevitability and potential of the Asterisk market and modify zap or make drivers for their equipment. This is a problem for independent developers because if you began to make this type of equipment, you have to potential of instant heavy competition. Maybe it would be a good open hardware project like zappata. Anyway, if you think asterisk is growing fast now, just wait till this happens... Anyway, I need hundreds of these phones. I am using Asterisk as a bridge to several aging PBXs and am expecting to one day run all of my wired phones native in Asterisk. John ___ 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] Analog FXO Woes Continue
I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ 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] Calls dropping, when server sysncs time?
Ok, I have had problems with calls dropping repeatedly today, does anyone have any suggestions on what to make sure is not running? I have x-windows disabled and Apache disabled. I noticed that mpg123 always seems to have 2 processes running, is there any way to drop this down to just 1? Also, can I stop the hald from running without any issues? I have looked at all my log files and I can not find any reason for why asterisk is dropping the calls. Thanks, Jared Armstrong ___ 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] Website that reads text recently on the list?
Thats it. Thanks! - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, December 07, 2004 9:52 AM Subject: Re: [Asterisk-Users] Website that reads text recently on the list? http://www.rhetorical.com/cgi-bin/demo.cgi Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. ___ 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] Budgetone 100 Caller ID
However when I dialled in and withheld my number, the Bt102 showed something which resembled ' tr1' ? That's its babytalk for asterisk! When we get calls with no CID, I do a setCallerID(000) for those phones hth ___ 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] High(er) availability
On Tue, 2004-12-07 at 09:18 -0600, Matthew Boehm wrote: This is a good question that the OP posted. Let say you have installed an Asterisk box at a customer location because they have 50 extensions and all talk to eachother alot. If their asterisk box fails, how can you re-direct them to your master box downtown? [snip] Don't know if I'm wording this right but one way would be to use a phone which sports 2 reg server entries in the config and a short refresh time. If the refrfesh of the reg on the primary server fails it will try to reregister with the secondary server down town. At least that's how I think it works. Regards, Patrick ___ 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] H.323 trunking
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with Q.931 cause)
-- Hungup 'OH323/L11490'
== No one is available to answer at this time
Dec 7 16:48:25 WARNING[1687569]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
what is the meaning of *reason 24*. Is there a problem
with my codec?
thx in advance...
ps: 61 is a prefix. All the extensions 61xxx
should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] Interrupt latency problems
Title: OpSign Have any of you tried to disable ACPI on the kernel? Rich Adamson wrote: On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote: Steven Critchfield wrote: On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote: So, isn't the issue he/I are chasing after essentially 'why is cpu consumption jumping 30% (or 100%) every ten seconds when zaptel is running with no calls present? So where is that CPU time going? Is it in the system, or userspace? Have you tried changing to a non FC or RH kernel as suggested earlier? Yes, I've just completed the installation of 2.6.9, and the spikes have gone away. Thank you, Steven. Your welcome. I am glad it solved the problem. Now if only someone knew what it was about the stock RH or FC kernel that makes it happen you could get RH or FC to stop using that patch. That or maybe more people will be like me and always cast a weary eye upon a prepackaged kernel no matter what distro it came from. Looking at the Changlog for 2.6.9, it would appear a fair amount of work has been down in the pci stuff and the interrupt support areas. Since that seems to be an issue that keeps rearing its head with the digium analog cards, maybe there is something 'fixed' in that area. Not being a strong linux admin, how difficult would you say installing 2.6.9 is on top of a RHv9 system (2.4.20-31.9) should be for me? Any suggestions/hints on how to do it would be appreciated. Rich ___ 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 -- __ AlessandroRen OpServices LucianadeAbreu,471-Sala403 PortoAlegre,RS-CEP90570-060 (phone55(51)3061-3588 4fax55(51)3061-3588 Qmobile55(51)9807-3255 :email[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] Analog FXO Woes Continue
[EMAIL PROTECTED] wrote: I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? You should probably scope the lines with a circuit tester. Used Wilcom T136 units can be had on eBay for about 20 bucks. They'll allow you to check the noise and loss on the circuit. When you report it you don't have to describe a problem, but simply state that the circuit is out of spec. No guarantee that this is your problem, but from the symptoms you describe you are definitely on the right track. Good luck. Jim. ___ 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] High(er) availability
Loosing calls wouldn't be to much of a problem I think, and it would be impossible to make a gracefull takeover if asterisk is in the mediastream. keepalived implements vrrp2 so that might be good enough. The problem lies in the registration data, but that could be solved by using fixed ip addresses for the phones. I need to setup a test environment, which I might just do :) Erik -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Tim Donahue Verzonden: dinsdag 7 december 2004 16:32 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] High(er) availability On Tue, 2004-12-07 at 15:47 +0100, Stefan de Konink wrote: E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Cisco claims that VRRP falls under one of their patents, so it could become an expensive option. There are several options out there at this point though that may be able to handle the needs for pre-empting the IP address. About 1 year ago the OpenBSD project wrote a patent-free alternative for VRRP called CARP. It allows for sharing of and automatic failover on an IP address. I have used it to build redundant firewalls that don't lose any state information when the connection drops. CARP is of course built into OpenBSD however I did find what looks to be a userland implementation for Linux. See www.ucarp.org for more information. There are other possible solutions as well, unfortunately I have not used any of these solutions they are just from brief google search. LVS (Linux Virtual Server) mentions VoIP services however I do not know if Asterisk would run in a cluster environment. There are also several sites that deal with high availibity from linux, the first one I noticed that looked like it had some really valuable information is www.linux-ha.org. Unfortunately this is all the easy part. The difficult part will be getting Asterisk to handle the failover gracefully. You probably don't want to lose all the SIP registration data and I have no idea if it will be possible to prevent you from losing the calls. You haven't named that as one of your goals, but it is always something to think about. -- Tim Donahue [EMAIL PROTECTED] Haynes Group, Incorporated ___ 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] astcc needs AGI.pm...where is it?
Greetings, I tried to build astcc, but the Makefile is looking for Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be and where it comes from? I've dragged in everything I can think of from cvs, and * is otherwise running fine. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ 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] Dropping calls, Polycom Renegotiation timeout?
Does anyone know if the renegotiation setting for the polycom phones will cause any existing/current calls to be dropped when the phone tries to renegotiate? I believe this might actually be what is causing my calls to be dropped. Like I said in my previous email I am not seeing any errors in my log file so I am hoping perhaps this is the issue. Thanks, Jared Armstrong ___ 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] Website that reads text recently on the list?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: December 7, 2004 9:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Website that reads text recently on the list? there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. Rhetorical http://www.rhetorical.com/cgi-bin/demo.cgi Enjoy, Jim. ___ 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] Fine Tuning
Hello all, We've been using our Asterisk system live for about a month now and I'm looking to tuning a few things. First, is echo, I receive a fair amount of echo during the first 10-15 seconds of incoming calls. Next is a very weird problem. We have serveral Polycom IP300's and one Budgetone phone. It seems that if we unplug move the Budgetone (which happens a fair amount as it is the phone we move to normally unreachable areas, some of the Polycoms will no longer work properly, the caller can hear us but we can't hear them. Very odd that the Budgetone triggers this but it is fairly consistent so we're close to just ditching the Budgetone. Sound quality, when the volume is maxed on our Polycom phones, the sound is jittery and you can here what sounds like some sort of artifacts from compression or something like that, in some of the worst cases you can even hear random little beeps in the background. Finally, all incoming calls seem to be a touch quite, basically the volume is always maxed on our Polycoms, is there a simple way to increase the incoming volume of all calls? If anyone can help or point me in the right direction on these, it would be much appreciated. I have gone through the wiki and not found much on these issues. Thanks, Pete ___ 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] Firewall traversal anomalies - AJA
I'm trying to setup a Cisco ATA 186 which has a public IP address but sits behind a firewall and connects to an Asterisk server with a NAT IP address sitting behind a BSD firewall. The Cisco registers with the Asterisk server without any problems, and I can place calls without any problems and the phone on the other end rings correctly. However, I cannot hear anything through the Cisco after the connection is made. Where should I begin looking for the problem? This is the sip.conf entry for the Cisco: [6184341501] callerid=GlobalEyes 6184341501 canreinvite=no context=from-internal dtmfmode=rfc2833 host=dynamic mailbox=x nat=yes port=5060 secret=xxx type=friend username=x allow=all You've picked _the_ most difficult of all configurations to get working (two nat's). You will likely hear about as many opinions about that on this list as their are active list members. There is no way for anyone to truly help you with this config unless you use a packet sniffer at various points to see exactly what is happening with the rtp port numbers and ip addresses. The reason for stating that is there are far too many variations in exactly how each firewall/nat box implements the nat function, and about as many variations in terms of what you are allowed to configured on each vendor's firewall. The bottom line is that you've apparently successfully map'ed the sip udp 5060 ports, but the voice is transported on rtp ports that are dynamically selected at the time the call is set up. If you look in /etc/asterisk/rtp.conf you'll see where asterisk selects from a large range of udp ports (for the rtp session). Each phone manufacturer has chosen their own range of rtp ports, and I've not seen two vendors actually use the same range. (Some phone vendors allow you to change that range while others don't.) So, when asterisk (as one example) begins the rtp setup (for audio), it might select udp port 12345, the phone might select 23456. If the nat boxes don't allow those two ports through (or if the nat box decides to map those ports to some other ports), the rtp session will never be established. Thus no audio. Even if you told us the exact model's of nat boxes you have installed, it won't do any good unless by chance someone in this world happens to have your exact same configuration. Not likely. So, _you_ really need to use a packet sniffer on both sides of your asterisk nat box and on both sides of your ata186 nat box to see what each of those boxes are doing to 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
[Asterisk-Users] Avaya 4606 IP Telephone
I have been trying to use the Avaya 4606 IP Telephone (with support for H323) with Asterisk. Has anyone else attempted this? Any success or definite failure? I know I must also use a gatekeeper with it, and I have tried both GNU Gatekeeper 2.0.8 and Open GK. However I have no success so far. The phone registers itself to the gatekeeper, and then unregisters. And this is repeated for as long as the phone is connected to the LAN. I know this is most probably related to the gatekeeper, but I would like to hear of any success or failure stories with the specific phone anyone else had. ___ 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] two questions
I see that the 100p is a modem with an Ambient chipset. Why does it sell for 80$ in some places? i can get Ambient pci modem down here for 9 dollars. Any difference? On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote: On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ 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] Analog FXO Woes Continue
Hi I feel your pain! We have had the same problem with our telco lines but found that converting to ISDN helped. If the delay on the send and receive two pair is to big the echo canceller is not strong enough. Try using a Voictronix card as they seem to solve the problem to a degree but I would suggest ISDN. Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Dugas Sent: Tuesday, December 07, 2004 5:32 PM To: Asterisk Mailing List Subject: [Asterisk-Users] Analog FXO Woes Continue I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad. Since this is obvioiusly a deal breaker, I'm looking for any more suggestions on how I might fet these things working. The hitch is pretty clearly the quality of the lines I have from BellSouth but I can't get thim to identify anything wrong. I have tried a Digium 1-port FXO card (can't remember part number and it's no longer on their site, hmmm...) as well as a Sipura SPA3000. With both of these interfaces, I'm getting consistent mis-dials on outbound calls, broken inbound fax-detection, broken DTMF detection in the attendant menus. Hours of adjustments to the gains on the Digium card only added echo and failed to reduce the offurenc of the other issues. These same two interfaces worked fine on a line at my office so I'm pretty sure the issue is with the lines at the test site. So, what are my options here for interfacing with these lines? Would the channel-bank route affect this? Thanks in advance for any suggestions, Paul -- Paul A. Dugas Dugas Enterprises, LLC email: [EMAIL PROTECTED]1711 Indian Ridge Drive phone: 404.932.1355 fax: 770.516-4841 Woodstock, GA 30189 USA [ onsite at the Georgia DOT's West Annex, 404.463.2860 x158 ] ___ 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] How to play messeage when user picks up the phone
Is it possible to play a message, when user pickups a phone. For example: press 1 to use this provider, press 2 to use this ... etc.. 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] two questions
Erick Perez wrote: I see that the 100p is a modem with an Ambient chipset. Why does it sell for 80$ in some places? i can get Ambient pci modem down here for 9 dollars. Any difference? Because Digium is selling support plus the modem, not just the modem. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ 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] Broadvoice patch and latest CVS version
Patch could not be applied to the latest cvs version and also http://www.voip-info.org/wiki-Asterisk+Broadvoice+patch?page=Asterisk%20Broadvoice%20patchcomments_threshold=0comments_offset=0comments_sort_mode=commentDate_desccomments_maxComments=10comments_parentId=1209#threadId1210 -- Best regards Vlad ___ 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] PRI errors
For a few weeks we have been getting errors that drop our PRI. The telco says the the line is clean and that our equipment is the problem. We're currently running Asterisk CVS-HEAD-12/03/04 but several versions have been tried in an attempt to fix the problem. The * server is based on a supermicro 1U chassis with a PIII 1.266GHz, 512MB RAM and a tor2 4port PCI card. The Tor2 card is on IRQ 11 which IS shared with an *unused* ethernet controller and USB port. It has worked like this for a long while. A single PRI is connected to port 1, a Microcom 4000/ISPorte is on port 2 and a Max 4000 is on port 3. We accept data and voice calls. When PRI drops, all calls are disconnected. If you happen to be on a voice call, you hear a brief PFFFT! as everything goes away. The line resyncs in a minute and everything operates normally until the next error. SIP -- SIP calls continue to work properly while the PRI is down. Our dialplan is extremely basic (too basic!) and has been in use since March/04. Until now the system has been very stable. The only time * has been down was when one of us botched the dialplan but that was found and fixed months ago. To save list bandwidth I'm linking the errors: http://www.linuxsys.com/files/pritrubbl.txt Before I tell the telco to come out I'd like a little insight to the error messages. Please copy replies to me directly. Thanks, -- Andrew McRory - President/CTO Linux Systems Engineers, Inc. - http://www.linuxsys.com Located in beautiful Tallahassee, Florida Office 850-224-5737 Office 850-575-7213 Mobile 850-294-7567 ___ 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] queue timeout
I'm trying to redirect the call to PSTN if no one is available in the queue or the agents in the queue do not answer. The following will redirect the call if no agents are logged in. But if the agent does not answer the call will timeout and the call will be terminated, not redirected. I've also tried to put the timeout rules in the queue context, no luck. [support] exten = 6040,1,Answer exten = 6040,2,Wait(1) exten = 6040,3,Playback(welcome) exten = 6040,4,Wait(1) exten = 6040,5,Queue(support|t|||1) exten = 6040,6,Dial(SIP/[EMAIL PROTECTED]) - call if no one answers [default] exten = t,1,Dial(SIP/[EMAIL PROTECTED]) Call queue 6040: -- Playing 'welcome' (language 'en') -- outgoing agentcall, to agent '6031', on 'Local/[EMAIL PROTECTED],1' -- Called Agent/6031 -- Called user -- Agent/6031 answered SIP/212.125.141.182-09b6c000 -- SIP/user-c2d5 is ringing -- Nobody picked up in 2 ms Call terminated. ___ 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] Question about e1/digium
There are some documentation about it ? Thanks Sergio Faulhaber [EMAIL PROTECTED] B. Vallet - www.acropolistelecom.net wrote: Yes it is possible but make distinct between simultaneous channels and phones numbers (DID) you can have for example 1000 phones numbers and 30 channels (E1) or 1 phone number and 30 channels. Benoit Vallet -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] De la part de SERGIO GUIMARAES FAULHABER Envoy: mardi 7 dcembre 2004 12:36 : [EMAIL PROTECTED] Objet: [Asterisk-Users] Question about e1/digium Hi all I am beginning in asterisk and am making tests with an ata-186. For the time being the tests are going well, however have a doubt. I am thinking about using a canal e1 with plate digium. Assuming that the company of telecommunications supplies e1 with 30 canals and numeration to me 4000-0001 4000-0029. she is possible to configure asterisk in way that somebody of is dials 4000-0025, to direct for a telephone sip ? Thanks for attencion Sergio Faulhaber [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 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] How to play messeage when user picks up the phone
I'm doing this in a call centre with Budgetone 100 telephones. But, in my case, its the Budgetones that offer the option to automatically dial an extension when the handset is lifted (or the speakerphone button is pressed) Derek PS. The latest release of the Budgetone firmware is broken and stops the auto-dial and message buttons from functioning correctly - stepping back a version fixes the problem. Bartosz Wegrzyn - asterisk wrote: Is it possible to play a message, when user pickups a phone. For example: press 1 to use this provider, press 2 to use this ... etc.. 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 -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.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] two questions
Christopher L. Wade schrieb: Because Digium is selling support plus the modem, not just the modem. But when you don't need the support? Bye! Michael ___ 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] Problem on Outgoing Calls (FXO - SIP)
Hello, We have a high volume of incoming and outgoing calls that come in via our analog POTS lines connected to FXO cards in an Adtran TA750. This is connected to a T100P. We are using Polycom IP 500's. The problem we are experiencing is, on frequent occasions, when someone dials out, there is another person who has just dialed in. I have had this problem at my house before (without an * system) where I pick up too quickly for it to ring. Are there any methods to mitigate this problem? I can see that Asterisk is recognizing the call, but it still opens a zap channel for the outgoing call. Any help would be appreciated. Thanks, Brent D. Franks Mindworks Internet Services ___ 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] Polycom IP 600 status setting in Asterisk
Hello! Am Mittwoch, 1. Dezember 2004 14:56 schrieb Michael Graves: I love my Polycom IP600s. However, I'm not clean on how the status setting on the phone impacts the behaviour of *. Anyone here have the details? No answers so far? Andi -- - Andreas Roedl- Senior IT Manager / Head of IT Dept. - NATIVE INSTRUMENTS GmbH - [EMAIL PROTECTED] - Schlesische Strasse 28 - http://www.native-instruments.de/ - D-10997 Berlin - Tel. +49-30-61 10 35-430 - Germany - Fax +49-30-61 10 35-35 ___ 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] IAX DIDs, Illinois
I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up multiple DIDs for different uses, but never having heard of these folks, I'm wondering what kind of a business I'm dealing with. Does anyone have experience with either of the above? I seem to remember recurring issues with VoicePulse... Does anyone know an ITSP that provides Illinois DIDs via IAX other than the two above? 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] High(er) availability
Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Cisco claims that VRRP falls under one of their patents, so it could become an expensive option. There are several options out there at this point though that may be able to handle the needs for pre-empting the IP address. About 1 year ago the OpenBSD project wrote a patent-free alternative for VRRP called CARP. It allows for sharing of and automatic failover on an IP address. I have used it to build redundant firewalls that don't lose any state information when the connection drops. CARP is of course built into OpenBSD however I did find what looks to be a userland implementation for Linux. See www.ucarp.org for more information. There are other possible solutions as well, unfortunately I have not used any of these solutions they are just from brief google search. LVS (Linux Virtual Server) mentions VoIP services however I do not know if Asterisk would run in a cluster environment. There are also several sites that deal with high availibity from linux, the first one I noticed that looked like it had some really valuable information is www.linux-ha.org. Unfortunately this is all the easy part. The difficult part will be getting Asterisk to handle the failover gracefully. You probably don't want to lose all the SIP registration data and I have no idea if it will be possible to prevent you from losing the calls. You haven't named that as one of your goals, but it is always something to think about. There is a lot of interest from lots of folks in how one handles failure overs, etc. I've got to believe that a fair number would be very happy with a primary-secondary arrangement where calls that were in-flight might be dropped, but recovery in terms of displacing the failed * box happens within several seconds (or possibly even a minute). Reading between the lines from the original poster's question, it would sound like that would be an acceptable aproach. If the failover time for a primary-secondary approach was short, keeping the registration data and other somewhat dynamic data in sync between boxes is relatively easy. It would seem the only remaining issues involve MAC addresses, redundant physical pstn-type interfaces (and probably something as simple as a relay flopping T1's or fxo's over). Has anyone truly implemented such a pri-sec failure, and if so, care to offer up some specific configuration data that is usable? ___ 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] two questions
Michael Vogel wrote: Christopher L. Wade schrieb: Because Digium is selling support plus the modem, not just the modem. But when you don't need the support? Bye! Michael Exactly. Choose the level of support you want from Digium and/or the list. Historically, Digium equipment gets support from both Digium and the list, non-Digium equipment only gets support from the list, and only when a valid reason for not using Digium exists. Please understand that Digium (Mark) is the reason * exists. Selling a $10 winmodem for $80 is the reason Digium exists. Digium's existence is the reason Mark can eat while he's coding for *. Mark being able to code * while he eats is the reason * is such a great project. Way to go Mark, et. al. (even the ones who don't get paid to eat while coding :) My $0.50. -Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ 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] Website that reads text recently on the list?
Nice sounding audio. On the demo there is a button to Download wav file This sounds like it should allow me to save the sample? Does not seem to work for me. thanks Mike On Tue, 07 Dec 2004 07:52:29 -0700, Darren Wiebe [EMAIL PROTECTED] wrote: http://www.rhetorical.com/cgi-bin/demo.cgi Darren Wiebe [EMAIL PROTECTED] Steve Totaro wrote: there was a website on the list recently that allowed you to enter text (up to 50 words) and it would create a wav file with various voice options. does anyone remember what it was? rapsody something or another. ___ 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
