Re: [Asterisk-Users] Who would use Asterisk SS7?
We would be a hour 0 user. And probably would also be abel to get some partners to test SS7 interconnect with since it would rid us of a hell of problems :) On Thursday 29 May 2003 2:22 pm, Mike M wrote: On Thursday 29 May 2003 05:27, Patrick wrote: On Thu, 2003-05-29 at 02:36, [EMAIL PROTECTED] wrote: On 24 May 2003, Thilo Salmon wrote: The other issue is a legal one. In order to connect to the incumbent telco your equipment has to be certified. I believe unless quite a few of us get together, this one might be a real problem. The SS7 equipment from Lucent, Nortel, Alcatel are likely already certified with the carrier you want to link to. They are. No worries about certs from those guys. If not, they are happy to make that happen for you. Also, in Europe you will not get an SS7 link to a carrier unless you are a licensed carrier yourself. True. But you will only be interested in SS7 if you are interested in being a licensed carrier and expanding to handle enough voice channels to make SS7 more cost effective than RBS. This point is at the heart of the original question. Putting SS7 on * is worthwhile only if there are going to be users. If SS7 were available today, would existing * users adopt SS7-IMT and would it interest non-users to become users? Easy solution -- Have * talk to SS7-certified equipment. Cisco comes to mind. They have SS7 gateways that could talk with * as do many others. You can use * to cut out the expensive hardware and only use the bare minimum of the vendor's setup to talk to SS7. -Dan Whatever * is able to cut out, you still need a serious telco budget to actually get the SS7 solution. Given customer requirements, you pass the $500,000 mark in the blink of an eye. And that does not include a service contract for the kit for as long as it is in service. The cost of traditional SS7 equipment is prohibitive for big and small business plans. A low-cost alternative could be a business plan enabler. This may still make sense to some though. If I were to make such investments I would: * become a licensed carrier * install SS7 interconnection gear with all major carriers in the designated area In North America you can connect to a single SS7 network provider and have all the SS7 access you need. SS7 access is separate from IMT access. I would think that connection to a single carrier in Europe would be sufficient to begin with also. * negotiate termination service fees as high as possible With your clients? * get tons of traffic to my network by offering ??? to customers For * I think an attraction is VoIP to PSTN bridging and access to the PSTN user base. This is a technology list so marketing ideas are OT. * profit! The dream of all operators :-). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming calls using iconnecthere
When an incoming call is attempted Asterisk displays the following: Warning [131081]: chan_sip.c Line 1991: (__transmit_response): Unable to determine sequence number from '' I've been seeing this problem within the last little while (sorry I can't be more specific). This probably isn't going to help you if you're behind NAT but it seems that changing the register host from natrelay.deltathree.com to sipauth.deltathree.com fixed it for me. -- Luke -- Luke Howard | PADL Software Pty Ltd | www.padl.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'Zap'
Hi list,
I have the follow configuration:
===
extension.conf:
===
[pstn]
ignorepat = 0
exten = _0,1,Dial(${TRUNK}/${EXTEN:1})
[default]
exten = 120,1,Dial(IAX/[EMAIL PROTECTED])
include = pstn
But, when I dial from my gnophone something like 097991269, asterisk console returns
the fallow message:
NOTICE[245775]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of
type 'Zap'
Could anyone help me?
thanks in advance
Eduardo
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Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
Hi all, ISDN is not an option for me... I don't want to subscribe to ISDN services just to use Asterisk. I have just tested a Cisco 1700 router as a 2 x FXS + 2 x FX interfaces. It is connected to Asterisk through ethernet. I have connected 2 phone lines and two analog phones. The box is seen by Asterisk as a SIP UA. I can call PSTN using separate prefixes for each of the two lines. The two analog phones are defined as extensions in Asterisk. I have some other phones connected to the same Asterisk: one Cisco 7960 and one X-Lite, both on SIP and a Netmeeting as a H.323 client. It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan - Original Message - From: Markku Korpi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 4:50 PM Subject: RE: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? If you can get ISDN, AVM has FRITZ!Card PCMCIA that works with * and CAPI. Markku -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dan Sent: Thursday, May 29, 2003 10:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? Hi, There is any external FXO device (1 or 2 lines only) , USB or Ethernet based, supported by Asterisk? I ask this because my Asterisk PBX is installed on a notebook (for space reason), without any PCI slot available. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users www.mailfiler.com [MKO-GKCMAF] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer incomplete when MOH enabled
Title: Message Potential Bug? CSV as of yesterday. Scenario: When Music on Hold is enabled, initiating a transfer from a Cisco 7960 using its built in transfer function (either transfer or blind transfer) to an analog phone on a TDM10B, the transfer is not sucessfull. The analog phone rings, and the Cisco phone indicates that the transfer is completed. But on the analog phone there isjust emptiness, and the calling party continues to hear the on hold music. # transfers works fine, and the built-in transfer functions work fine when on hold music is not enabled.
Re: [Asterisk-Users] Who would use Asterisk SS7?
On Thursday 29 May 2003 09:38, Michael Bielicki wrote: We would be a hour 0 user. And probably would also be abel to get some partners to test SS7 interconnect with since it would rid us of a hell of problems :) :-) I've been following the 2-4 port T1 cards thread closely because that's the kind of application that could benefit from having SS7-IMT. -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1-PRI deployment questions...
I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten= definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who would use Asterisk SS7?
[EMAIL PROTECTED] writes: Easy solution -- Have * talk to SS7-certified equipment. Cisco comes to mind. have you checked the price of e.g. cisco sip/ss7 gw lately? i did a few months ago and it was huge. -- juha ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fault tolerance?
hi all what sort of fault tolerance (if any) exists for asterisk? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1-PRI deployment questions...
On Thu, 2003-05-29 at 10:04, John Harragin wrote: I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten= definitions? I'll answer this since I just had mine installed and went through this. Signalling comes in just a few options at the base level and is specific to the T1 transport. Your options are ESF or D4 for framing, and AMI or B8ZS for error detection. Your PRI rides on top of the T1 transport. In the PRI your DIDs are passed as part of the call setup in Q.931 packets on the D channel. Specifically you will place all your incoming PRI lines into a context in extentions.conf and then you will make extensions with the incoming phone numbers or part there of if your telco provider isn't sending the whole called number. In these extension deffinitions you can redirect the calls to either the appropriate IVR menu, or an internal extention. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
B8ZS/ESF I believe is the usual for a PRI DID calls in asterisk are routed just like dtmf dialed extensions, but there are not DTMF tones passed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Thursday, May 29, 2003 11:05 AM To: Asterisk Subject: [Asterisk-Users] T1-PRI deployment questions... I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten= definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
The Snom200's sourced from the UK are apparantly patched (hardware) so that the PC headset plugs will work. I haven't used the PC headset plugs, but a colleague using an RJ10 (call centre type) headset into the bottom, reports that it works ok. Tan - Original Message - From: Simon Woodhead [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 2:33 PM Subject: Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK? Hey Tan, I don't think was discussed in the past discussion so I'd appreciate your comments... I opted for a single snom 100 to test because we wanted to use the headset and I was told the 200 doesn't yet have headset support, despite the sockets being there. This is coming shortly apparently. Anyway, the support for the headset on the 100 is really bugging me. You can dial off-hook and then press a button to switch from speaker to headset but as soon as the call is answered it reverts and you have someone shouting hello across the office. We want users to use the handset for infrequent use but the headset more commonly and in an open-plan office don't want the first few seconds of every call being broadcast. Am I missing a setting on the 100 to make the headset selection more permanent? Does the 200 support headsets after all? Properly? Many thanks, Simon - Original Message - From: T Aksoy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 1:13 PM Subject: Re: [Asterisk-Users] What is the going rate for the Snom 100 in the UK? Our prices (for singles): SNOM 100: £169+VAT (free delivery) SNOM 200: £189+VAT (free deliver) Personally (as I'm sure the guys on this group would agree) I would go for the SNOM 200. Look on the emailing list for a lengthy discussion on this subject. Tan - Original Message - From: nathan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 12:44 PM Subject: [Asterisk-Users] What is the going rate for the Snom 100 in the UK? Hi All, What is the going rate for the Snom 100 in the UK? I've found a couple of suppliers with prices around the £170 (exc vat) mark. Regards, Nathan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] What is the going rate for the Snom 100 in the UK?
Our prices (for singles): SNOM 100: £169+VAT (free delivery) SNOM 200: £189+VAT (free deliver) Personally (as I'm sure the guys on this group would agree) I would go for the SNOM 200. Look on the emailing list for a lengthy discussion on this subject. Tan Do you have a website? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote: B8ZS is required for PRI. It's a digital service and can not handle the loss of data required for AMI. I wasn't aware that AMI lost data. AMI just inverts polarity on the line for every other 1. B8ZS does the same thing but intentionally introduces errors on the line to maintain 1's density. Neither one is lossy. On Thu, 29 May 2003, Joe Antkowiak wrote: B8ZS/ESF I believe is the usual for a PRI DID calls in asterisk are routed just like dtmf dialed extensions, but there are not DTMF tones passed. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Harragin Sent: Thursday, May 29, 2003 11:05 AM To: Asterisk Subject: [Asterisk-Users] T1-PRI deployment questions... I am ordering T1-PRI service from local service provider and have a few questions. Is there framing and coding considerations (or is it all one standard), if so what is best? How are calls routed based on DIDs - are these just dtmf tones passed after the call is picked up and treated as normal exten= definitions? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACD
Good day, Our installation needs a robust ACD application (as I'm sure others do) that can be dynamically reconfigured (if possible) maybe by a MySQL database. I have looked at Bill Heckel's ACD work and Andreas Otto's DynExtendb as well as James Sharp's ACD. None of these seem to be quite finished. Is there a chance a unified (* blessed) ACD application can be put together by those who know how? My knowledge of the Asterisk API is next to zero and the time to learn it would be prohibitivly long much less relearning C. Multiple groups with overlapping agents are a requirement as is roundrobin (per group), next available, and longest time since last call answered (per group). Anybody's input on this would be appreciated. Asterisk has worked out fine in our test environment and we are needing this for deployment to replace our existing proprietary system. Thanks for your time. Jim Friedeck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 codecs not allowing * as deamon ?
Try running asterisk like this: screen -d -m asterisk -vvvc or screen -d -m asterisk -c or screen -d -m asterisk -f Martin On Thu, 29 May 2003, Tjardick van der Kraan wrote: When we have the G.729 codec (ordered from digium) active in * we have the following problem: running * in standard deamon mode: asterisk starts * and then when done loading it ends. when loading * in console mode: asterisk -c it runs just fine. also running *: asterisk -v makes it run properly. When we disable G.729 from the modules it just runs fine again in standard demon mode. Is this a bug or is it this way on purpose. If so, why ? Greetings, Tj -- Tjardick van der Kraan [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Do you have your zap channel in asterisk when you type zap show channels
?
If not than make sure you have a proper config files (zaptel.conf
zapata.conf)
Martin
On Thu, 29 May 2003, Eduardo Goncalves wrote:
Hi list,
I have the follow configuration:
===
extension.conf:
===
[pstn]
ignorepat = 0
exten = _0,1,Dial(${TRUNK}/${EXTEN:1})
[default]
exten = 120,1,Dial(IAX/[EMAIL PROTECTED])
include = pstn
But, when I dial from my gnophone something like 097991269, asterisk console returns
the fallow message:
NOTICE[245775]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of
type 'Zap'
Could anyone help me?
thanks in advance
Eduardo
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Re: [Asterisk-Users] Setting up fax on *
Lets say that your E1 channels are assinged to context=incoming channel = 1-15,17-31 Then in extensions.conf in context [incoming] exten = fax,1,Dial,Zap/1 ;if your Zap/1 port is FXS where the fax ;is attached (all other extensions) regards Martin On Thu, 29 May 2003 [EMAIL PROTECTED] wrote: Hello All, I am using an E100P card on a PRI line. I need to setup a FAX extension. Can somebody help me please? Marco ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] a beginner's SIP question ..
I am trying to get asterisk to dial this address
:
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED] --
SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6 --
Attempting native bridge of SIP/sipphone-97b6 and
SIP/216.52.153.207-7c3b
but on my headset all I get is silence .. the call
doesn't drop though.
What am I doing wrong ?
many thanks,
Dave
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
Hi, I look for something in the price range of a X100P for one FXO port. regarding the Dlink device, I think that there is not a real FXO port, more somethink like in Actiontec's InternetPhoneWizard, just to be able to use the analog phones for both IP and PSTN calls. It just switch one of the phone t the PSTN line. Have you considered a S100U and one of those $35 FXS to FXO converters? There is something like that? Where I can find such a converter and how this thing works? BR, Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 6:34 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CalledID by channel difficulties
Ok, I want to be able to set a different callerid for each Zapata channel. -[zapata.conf]- callerid=Reception 0 channel=3 callerid=Batman 2000 channel=4 callerid=Robin 1001 channel=5 callerid=The Joker 1002 channel=6 group=2 channel=3-6 ;TDM10B Whenever I dial an extension, the callerid only shows the last defined callerid (The Joker / 1002) Why is this happening? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] a beginner's SIP question ..
Hi,
Check to have a common set of codecs.
If X-Lite is used and at the other end is a phone
without GSM support, then it doesn't work.
Try to disable GSM on the soft phone (if
X-Lite).
BR,
Dan
- Original Message -
From:
Dave Alan Caruana
To: [EMAIL PROTECTED]
Sent: Thursday, May 29, 2003 9:01
PM
Subject: [Asterisk-Users] a beginner's
SIP question ..
I am trying to get asterisk to dial this address
:
sip:[EMAIL PROTECTED]
Using a softphone on my PC
(217.168.168.49)
it dials immediately and I get a voice prompt
..
I have configured an extension, 1303 on
asterisk,
modifying the demo configuration :
exten = 1303,1,Dial(SIP/[EMAIL PROTECTED])
When from my softphone I dial
sip:[EMAIL PROTECTED]
on the console I get :
-- Executing
Dial("SIP/sipphone-97b6", "SIP/[EMAIL PROTECTED]") in new
stack -- Called [EMAIL PROTECTED]
-- SIP/216.52.153.207-7c3b answered SIP/sipphone-97b6 --
Attempting native bridge of SIP/sipphone-97b6 and
SIP/216.52.153.207-7c3b
but on my headset all I get is silence .. the
call doesn't drop though.
What am I doing wrong ?
many thanks,
Dave
[Asterisk-Users] Outbound calls bridging
Hi All, With the help and patience of this forum, I have been able to set my asterisk box to make outbound calls to iconnecthere. My intention is to make two such calls and bridge them( three way calling) . Based on a earlier suggestion, I have created two accounts with iconnect and have successfully registered both the accounts from asterisk. I am having difficulty in setting my extension.conf to make two calls on the iconnect channels and brige them. Any help is greatly appreciated. If there is another way to do this please let me know. Thanks Pradeep _ Watch Hallmark. Enjoy cool movies http://server1.msn.co.in/sp03/hallmark/index.asp Win hot prizes! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CalledID by channel difficulties
On Thu, 2003-05-29 at 13:17, Derek Beaumont wrote: Ok, I want to be able to set a different callerid for each Zapata channel. -[zapata.conf]- callerid=Reception 0 channel=3 callerid=Batman 2000 channel=4 callerid=Robin 1001 channel=5 callerid=The Joker 1002 channel=6 group=2 channel=3-6 ;TDM10B Whenever I dial an extension, the callerid only shows the last defined callerid (The Joker / 1002) Why is this happening? Move the group=2 above your other channel definitions, and remove the channel=3-6 line. The resaon your getting just the last caller ID is because you redefine the channels with only the last callerID. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk IAX over VSAT satellite.
Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The other end (gateway) is a P200MMX with a X100P FXO card. I have bi-directional calling set up so that the VSAT-phone can make outbound calls using the X100P in the gateway, and if the X100P gets a ring it answers and transfers the call to the analog phone on the other side of the VSAT. There is about a 1-2 second propagation delay in voice from the VSAT phone, as expected. The echo is not bad at all, and the voice quality is quite good. I don't think the VSAT network was very busy so I don't know how well this will work if the available bandwidth is less. We are not using the G.729 codec - just gsm. I have tos=reliable set in iax.conf. I didn't get disconnected during my test calls, but they weren't very long in duration. I haven't tried a fax but maybe I will. Anyway congratulations Mark et al on your fine work making such a robust VoIP system. Thanks! -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CalledID by channel difficulties
The zapata.conf file is parsed from the top down, so Asterisk uses the value since the last channel keyword. so Asterisk really only sees: callerid=The Joker 1002 group=2 channel=3-6 ;TDM10B in your zapata.conf file Jeremy McNamara Derek Beaumont wrote: Ok, I want to be able to set a different callerid for each Zapata channel. -[zapata.conf]- callerid=Reception 0 channel=3 callerid=Batman 2000 channel=4 callerid=Robin 1001 channel=5 callerid=The Joker 1002 channel=6 group=2 channel=3-6 ;TDM10B Whenever I dial an extension, the callerid only shows the last defined callerid (The Joker / 1002) Why is this happening? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Then propably your board stoped taking interrupts. Try changing the PCI slot or IRQ. Make sure you don't run X-windows. regards Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 11:41:01 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Do you have your zap channel in asterisk when you type zap show channels ? If not than make sure you have a proper config files (zaptel.conf zapata.conf) Martin Yes, I do *CLI zap show channels Chan. Num. Extension ContextLanguage MusicOnH 1default 2default 3default 4default *CLI === zaptel.conf === #EM span=1,0,0,cas,hdb3 em=1-4 loadzone = us defaultzone=us zapata.conf [channels] group = 1 context=default signalling=em channel = 1-4 usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=yes thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk IAX over VSAT satellite.
What bandwidth do you have available for you connection (upsteram and downstream)? Do you have any CIR for VSAT connection ? Martin On Thu, 29 May 2003, Jim Ockers wrote: Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The other end (gateway) is a P200MMX with a X100P FXO card. I have bi-directional calling set up so that the VSAT-phone can make outbound calls using the X100P in the gateway, and if the X100P gets a ring it answers and transfers the call to the analog phone on the other side of the VSAT. There is about a 1-2 second propagation delay in voice from the VSAT phone, as expected. The echo is not bad at all, and the voice quality is quite good. I don't think the VSAT network was very busy so I don't know how well this will work if the available bandwidth is less. We are not using the G.729 codec - just gsm. I have tos=reliable set in iax.conf. I didn't get disconnected during my test calls, but they weren't very long in duration. I haven't tried a fax but maybe I will. Anyway congratulations Mark et al on your fine work making such a robust VoIP system. Thanks! -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 12:08:32 -0700 Andrew Gillham [EMAIL PROTECTED] wrote: Does it work without the group? e.g. Zap/1 Also, does 'zap show channel 1' look ok? -Andrew yeap, I tried Zap/1 and it didn't work. :~( *CLI zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Context: default Caller ID string: Destroy: 0 Signalling Type: E M Immediate Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No *CLI thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 2003-05-29 at 14:27, Eduardo Goncalves wrote: On Thu, 29 May 2003 14:08:01 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Then propably your board stoped taking interrupts. Try changing the PCI slot or IRQ. Make sure you don't run X-windows. My box has only one slot. I changed the IRQ, but still get the same error. What MB are you using, and what chipset is on it? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 14:08:01 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Then propably your board stoped taking interrupts. Try changing the PCI slot or IRQ. Make sure you don't run X-windows. My box has only one slot. I changed the IRQ, but still get the same error. Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANI matching trouble
On 2003-05-28 at 22:39, Mark Spencer ([EMAIL PROTECTED]) wrote: exten = 4044633/_213.,1,OurApp,losangeles-queue exten = 4044633/_.,1,OurApp,default-queue Take out the _. rule and just leave it 4044633 and it should work fine. That did it. Works great! Thanks. Not postive the _ is required on matching the callerid part, but honestly i just don't remember. It _is_ required. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Martin The command strace -xx cat /dev/zap/1 didn't stop here my /proc/interrupts asterisk:~# cat /proc/interrupts CPU0 0: 114109 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 3:1083355 XT-PIC tor2 11: 59 XT-PIC cmpci 12: 7962 XT-PIC eth0 14: 2495 XT-PIC ide0 NMI: 0 LOC: 114078 ERR: 0 thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk IAX over VSAT satellite.
Hey Jim, All sounds good. We tried a satellite system here a few months ago but couldn't get on with it. Glad you've had more success. In theory, it shouldn't matter whether the TCP/IP link between your sites is going over satellite, modem or any other medium but the issues we found with satellite that would be particularly damaging for VoIP were as follows: - Latency. You're onto this one already by the sounds of it. We were seeing 750ms pings so you're looking at delays of around 1 second; 1.5-2 seconds for someone to hear what you've said and reply. That doesn't prevent a conversation but might make it sound a little strange to the other party who doesn't know what is going on. - Upstream. We had a system with 2Mbps downstream but since the upstream is the expensive part for providers to provide it is usually much much smaller - ours was only 128k. That is one call for many codecs without allowing for any other use you'll be making of the line. G.729 would improve this a lot as you've spotted. - Drop-outs. A satellite system should theoretically provide continuous service like a leased line or modem connection so you shouldn't get call dropouts. However, we found that we'd lose all connectivity from our provider for several seconds at a time. It could have been a peculiarity of the way they were prioritising traffic, routing, excessive contention or even the non-TCP/IP method for the dishsatdish part of the link but it seems whenever other customers were making heavy downloads others would slow down to just a few bps or drop out completely. That wouldn't be good for the quality of any calls in progress even if the connection was maintained. I'm not meaning to be negative or dash your enthusiasm but if I had a choice of links to do VoIP over, satellite would be at the bottom, even below modems. Our experience could be unique of course and if you own both ends of the link then you have far more control over the issues I've mentioned, other than latency of course. All the best, Simon - Original Message - From: Jim Ockers [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 7:47 PM Subject: [Asterisk-Users] Asterisk IAX over VSAT satellite. Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The other end (gateway) is a P200MMX with a X100P FXO card. I have bi-directional calling set up so that the VSAT-phone can make outbound calls using the X100P in the gateway, and if the X100P gets a ring it answers and transfers the call to the analog phone on the other side of the VSAT. There is about a 1-2 second propagation delay in voice from the VSAT phone, as expected. The echo is not bad at all, and the voice quality is quite good. I don't think the VSAT network was very busy so I don't know how well this will work if the available bandwidth is less. We are not using the G.729 codec - just gsm. I have tos=reliable set in iax.conf. I didn't get disconnected during my test calls, but they weren't very long in duration. I haven't tried a fax but maybe I will. Anyway congratulations Mark et al on your fine work making such a robust VoIP system. Thanks! -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On 29 May 2003 14:32:01 -0500 Steven Critchfield [EMAIL PROTECTED] wrote: What MB are you using, and what chipset is on it? Silicon Integrated Systems [SiS] 620 Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
So now that I finally relize that you're using T1 or E1 circuit Do you have a ISDN PRI or an analog ciruit ? What's the status of the span in zttool or in (/proc/zaptel/1). Is it OK, RED, YELLOW ? Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Martin The command strace -xx cat /dev/zap/1 didn't stop here my /proc/interrupts asterisk:~# cat /proc/interrupts CPU0 0: 114109 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 3:1083355 XT-PIC tor2 11: 59 XT-PIC cmpci 12: 7962 XT-PIC eth0 14: 2495 XT-PIC ide0 NMI: 0 LOC: 114078 ERR: 0 thanks Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Sorry Martin, I checked the strace output and it stoped with some messages, like this: open(/dev/zap/1, O_RDONLY|O_LARGEFILE) = -1 EBUSY (Device or resource busy) Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
Didn't you just write a post before that it was running ? The EBUSY means that you propably have asterisk running and the port is busy or you have strace line on some other console Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 14:32:37 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Check whether strace -xx cat /dev/zap/1 gives you any output If it stops and waits than your board is not taking interrupts. Is the board running on the separate IRQ ?(/proc/interrupts) Sorry Martin, I checked the strace output and it stoped with some messages, like this: open(/dev/zap/1, O_RDONLY|O_LARGEFILE) = -1 EBUSY (Device or resource busy) Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
On 29 May 2003, Steven Critchfield wrote: On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote: B8ZS is required for PRI. It's a digital service and can not handle the loss of data required for AMI. I wasn't aware that AMI lost data. AMI just inverts polarity on the line for every other 1. B8ZS does the same thing but intentionally introduces errors on the line to maintain 1's density. Neither one is lossy. AMI is lossy. When the ones density in the signal is too low, AMI will insert ones to ensure that the far end does not lose sync. C. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 14:58:09 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: So now that I finally relize that you're using T1 or E1 circuit Do you have a ISDN PRI or an analog ciruit ? What's the status of the span in zttool or in (/proc/zaptel/1). Is it OK, RED, YELLOW ? Martin It's an E1 circuit with four channels, EM immediate signalling. I dont have ISDN neither an analog The alarm is OK Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk IAX over VSAT satellite.
Martin, What bandwidth do you have available for you connection (upstream and downstream)? Do you have any CIR for VSAT connection ? I think we have 400Kbps downstream and 56-112Kbps upstream. No CIR that I know of, it's first come first served for the bandwidth, and it's all shared all the time as far as I know. However it's pretty good broadband service and definitely usable for interactive stuff like webex or ssh, so I guess I shouldn't be surprised that it works OK for VoIP as well. -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
So it means that the board is working all right but there is problem with the telco or you're using diffrent signalling for your circuit. Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 15:06:12 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: Didn't you just write a post before that it was running ? The EBUSY means that you propably have asterisk running and the port is busy or you have strace line on some other console Martin yes, asterisk was running when strace showed this message with asterisk not running, strace didnt stoped to show a lot of chars ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Examples of using console as normal channel?
I would like to take advantage of my soundcard/OSS system but so far haven't come on to examples of what the specs would look like, particularly for bridging a call onto the console. Also I wonder whether the kernel version of OSS works all right for this, as opposed to the official OSS commercial product. Thanks in advance for any pointers that might be forthcoming. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote: On 29 May 2003, Steven Critchfield wrote: On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote: B8ZS is required for PRI. It's a digital service and can not handle the loss of data required for AMI. I wasn't aware that AMI lost data. AMI just inverts polarity on the line for every other 1. B8ZS does the same thing but intentionally introduces errors on the line to maintain 1's density. Neither one is lossy. AMI is lossy. When the ones density in the signal is too low, AMI will insert ones to ensure that the far end does not lose sync. As I understand it, AMI is not lossy but will may cause problems due to not maintaining 1's density. 1's density is used to make sure both sides are synced up properly. B8ZS is AMI except that it introduces bipolar violoations to make sure the line doesn't stay in an off state for too long. So B8ZS intentionally throws errors on the line in a known manner so as to make sure each side is in sync, but AMI does not care if the line goes all 0's for a while. I still haven't been able to dig up any documentation to back up that AMI is lossy, just maybe prone to errors via slips. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 15:26:25 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: So it means that the board is working all right but there is problem with the telco or you're using diffrent signalling for your circuit. Martin I've just called my telephony provider and reliaze that the zaptel's signaling bits was inverted. The provider adjusted his bits and I could make a call. by the way, how can I configure the signaling bits? thanks for the help Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1-PRI deployment questions...
On Thu, 2003-05-29 at 15:46, Steven Critchfield wrote: On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote: On 29 May 2003, Steven Critchfield wrote: On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote: B8ZS is required for PRI. It's a digital service and can not handle the loss of data required for AMI. I wasn't aware that AMI lost data. AMI just inverts polarity on the line for every other 1. B8ZS does the same thing but intentionally introduces errors on the line to maintain 1's density. Neither one is lossy. AMI is lossy. When the ones density in the signal is too low, AMI will insert ones to ensure that the far end does not lose sync. As I understand it, AMI is not lossy but will may cause problems due to not maintaining 1's density. 1's density is used to make sure both sides are synced up properly. B8ZS is AMI except that it introduces bipolar violoations to make sure the line doesn't stay in an off state for too long. So B8ZS intentionally throws errors on the line in a known manner so as to make sure each side is in sync, but AMI does not care if the line goes all 0's for a while. I still haven't been able to dig up any documentation to back up that AMI is lossy, just maybe prone to errors via slips. I still can't find any reference to AMI being lossy, and can't find any comments that show where a AMI circuit would introduce 1's to maintain 1's density. After reading a page describing test patterns and why they use certain test patterns, it makes sense why AMI might not be usable for a PRI though. http://www.electrodata.com/testpat.htm In a PRI, since the signalling is in the D channel, and the consecutive B channels could be completely clear, you could run into times with more than 15 consecutive zeros. Although I need to do more looking at how D4 or ESF lays on top of a T1 signal. Anyways, with more than 15 consectuives zeros you no longer are within ANSI spec. On a RBS circuit it would be less likely to fall too far out of spec using AMI. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk IAX over VSAT satellite.
Remember that a ping is round trip so the other user should only experience a 325ms delay on a 650ms circuit. What you would be expieriecing is the overlap in conversations as a result of the delay. i.e. when someone stops talking, it takes about 300ms for the other side to start getting the pause and then 300 more ms for the persons responce to get back to the original speaker. While there are many other pitfalls to VoIP and TCP-IP over satellite (I can expond on many of them as I work for a company in the buisiness) it still is a viable option in many situations. Mobile sites, remote locations, maritime, quick responce and many other unique situations make satellite a very viable option for many users. As far as dropouts are concerned, a well designed satellite system can operate at 99.995% availability or more. For licencing concerns, Ku is prefered over C. Most systems sold as consumer systems are also sharing bandwidth which would cause some additional concerns for the importance of the data being transported. (I can expound if anyone is interested.) I have a testbed up here constantly and can go for about a month before noticing any downtime. Chad C. Wicker Systems Engineer Petrocom [EMAIL PROTECTED] 5/29/2003 2:39:16 PM Hey Jim, All sounds good. We tried a satellite system here a few months ago but couldn't get on with it. Glad you've had more success. In theory, it shouldn't matter whether the TCP/IP link between your sites is going over satellite, modem or any other medium but the issues we found with satellite that would be particularly damaging for VoIP were as follows: - Latency. You're onto this one already by the sounds of it. We were seeing 750ms pings so you're looking at delays of around 1 second; 1.5-2 seconds for someone to hear what you've said and reply. That doesn't prevent a conversation but might make it sound a little strange to the other party who doesn't know what is going on. - Upstream. We had a system with 2Mbps downstream but since the upstream is the expensive part for providers to provide it is usually much much smaller - ours was only 128k. That is one call for many codecs without allowing for any other use you'll be making of the line. G.729 would improve this a lot as you've spotted. - Drop-outs. A satellite system should theoretically provide continuous service like a leased line or modem connection so you shouldn't get call dropouts. However, we found that we'd lose all connectivity from our provider for several seconds at a time. It could have been a peculiarity of the way they were prioritising traffic, routing, excessive contention or even the non-TCP/IP method for the dishsatdish part of the link but it seems whenever other customers were making heavy downloads others would slow down to just a few bps or drop out completely. That wouldn't be good for the quality of any calls in progress even if the connection was maintained. I'm not meaning to be negative or dash your enthusiasm but if I had a choice of links to do VoIP over, satellite would be at the bottom, even below modems. Our experience could be unique of course and if you own both ends of the link then you have far more control over the issues I've mentioned, other than latency of course. All the best, Simon - Original Message - From: Jim Ockers [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 7:47 PM Subject: [Asterisk-Users] Asterisk IAX over VSAT satellite. Hi all, For some reason VSAT or Satellite Internet services are not mentioned (or searchable) in this list's archives. I thought I'd let you know that I tested Asterisk using IAX (not IAX2) to make a phone call from an analog phone hooked up to an Asterisk system behind a Linksys router connected to a Gilat VSAT satmodem, and it worked. The other end (gateway) is a P200MMX with a X100P FXO card. I have bi-directional calling set up so that the VSAT-phone can make outbound calls using the X100P in the gateway, and if the X100P gets a ring it answers and transfers the call to the analog phone on the other side of the VSAT. There is about a 1-2 second propagation delay in voice from the VSAT phone, as expected. The echo is not bad at all, and the voice quality is quite good. I don't think the VSAT network was very busy so I don't know how well this will work if the available bandwidth is less. We are not using the G.729 codec - just gsm. I have tos=reliable set in iax.conf. I didn't get disconnected during my test calls, but they weren't very long in duration. I haven't tried a fax but maybe I will. Anyway congratulations Mark et al on your fine work making such a robust VoIP system. Thanks! -- Jim Ockers, P.Eng. ([EMAIL PROTECTED]) Contact info: please see http://www.ockers.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
I think they are hardcoded. But what do you exactly refer to by signalling bits ? Martin On Thu, 29 May 2003, Eduardo Goncalves wrote: On Thu, 29 May 2003 15:26:25 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: So it means that the board is working all right but there is problem with the telco or you're using diffrent signalling for your circuit. Martin I've just called my telephony provider and reliaze that the zaptel's signaling bits was inverted. The provider adjusted his bits and I could make a call. by the way, how can I configure the signaling bits? thanks for the help Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
I think this is the company that makes them but it is hard to tell. http://www.artech.com.tw/html/english/AX300/AX300.htm This company sells them http://www.aislecom.com/ A rep. for them posted this thread, claimed to be the manufacturer. http://lists.digium.com/pipermail/asterisk-users/2003-March/009134.html There are quite a few comments so click on Next Message There is someone on eBay selling them http://tinyurl.com/bp4x disclaimer: I have never used one. I am not associated with the seller. On other lists I did hear some people had problems with them. You may want to start another thread and ask if any * users are using them. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 2:10 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? Hi, I look for something in the price range of a X100P for one FXO port. regarding the Dlink device, I think that there is not a real FXO port, more somethink like in Actiontec's InternetPhoneWizard, just to be able to use the analog phones for both IP and PSTN calls. It just switch one of the phone t the PSTN line. Have you considered a S100U and one of those $35 FXS to FXO converters? There is something like that? Where I can find such a converter and how this thing works? BR, Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 6:34 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channel of type 'Zap'
On Thu, 29 May 2003 16:16:29 -0500 (CDT) Martin Pycko [EMAIL PROTECTED] wrote: I think they are hardcoded. But what do you exactly refer to by signalling bits ? Martin Bits To tell the status of a channel. It's four (ABCD) Transmit/Receive signaling bit patterns for the Idle and Seized states. You can read details here: http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00801123bb.shtml [ ]'s Eduardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Issue with connected TA 750
Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a bum BCU? Also, when performing a ring test from the admin port of the 750, the same behavior is present. Any ideas on this one? Thanks in advance. Scott Bisker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Issue with connected TA 750
you sure you don't have a multiplying block (I use the nordx stuff but I am sure there is an equivalent on every manufacturer's stuff) did this once accidentally, and was so pissed when I realized the problem, I made sure that block would never get used again. I check the product id every time now before I punch down anything. At 06:58 PM 5/29/2003 -0400, you wrote: Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a bum BCU? Also, when performing a ring test from the admin port of the 750, the same behavior is present. Any ideas on this one? Thanks in advance. Scott Bisker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1-PRI deployment questions...
Steven Critchfield wrote: On Thu, 2003-05-29 at 15:46, Steven Critchfield wrote: On Thu, 2003-05-29 at 15:06, Charles E. Youse wrote: On 29 May 2003, Steven Critchfield wrote: On Thu, 2003-05-29 at 10:44, Charles E. Youse wrote: B8ZS is required for PRI. It's a digital service and can not handle the loss of data required for AMI. I wasn't aware that AMI lost data. AMI just inverts polarity on the line for every other 1. B8ZS does the same thing but intentionally introduces errors on the line to maintain 1's density. Neither one is lossy. AMI is lossy. When the ones density in the signal is too low, AMI will insert ones to ensure that the far end does not lose sync. As I understand it, AMI is not lossy but will may cause problems due to not maintaining 1's density. 1's density is used to make sure both sides are synced up properly. B8ZS is AMI except that it introduces bipolar violoations to make sure the line doesn't stay in an off state for too long. So B8ZS intentionally throws errors on the line in a known manner so as to make sure each side is in sync, but AMI does not care if the line goes all 0's for a while. I still haven't been able to dig up any documentation to back up that AMI is lossy, just maybe prone to errors via slips. I still can't find any reference to AMI being lossy, and can't find any comments that show where a AMI circuit would introduce 1's to maintain 1's density. After reading a page describing test patterns and why they use certain test patterns, it makes sense why AMI might not be usable for a PRI though. http://www.electrodata.com/testpat.htm In a PRI, since the signalling is in the D channel, and the consecutive B channels could be completely clear, you could run into times with more than 15 consecutive zeros. Although I need to do more looking at how D4 or ESF lays on top of a T1 signal. Anyways, with more than 15 consectuives zeros you no longer are within ANSI spec. On a RBS circuit it would be less likely to fall too far out of spec using AMI. You are right, Steve. AMI isn't lossy. It stands for alternate mark inversion. It simply forces more transitions into the stream to ensure good sync at the receiver. With the way old T1s worked this was good enough to ensure sync., as the content of these T1s was always voice, and not completely arbitrary. With ISDN, or other data applications, which carry completely arbitrary content as well as voice, you really need a more robust sync scheme. This caused changes in both E1s and T1s. E1s got CRC4 to ensure robust frame sync. T1s got 8BZS to ensure robust bit sync. An ISDN T1 *should* be using 8BZS, but isn't always. Similarly, an ISDN E1 *should* be using CRC4 framing, but some countries insist on the older flakier framing mechanism for some odd reason. I think the confusion about lossy T1s probably relates to the robbed bit signalling that is generally used on non-ISDN T1s. This does, of course, lose a little of the 64kbps channel in a rather PITA way. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] aastra pt480 and adsi
Ok, so I figured out my problem with my pt480s. But, now I have a few more. 1. When I dial into the voicemailmain or voicemailmain2 application, the phone and * start talking adsi, but then the phone tells me programming download canceled, services is full., but my services list isn't full, only Asterisk PBX occupies slot 2, slots 1, 3 and 4 are available. Any ideas? I have tried erasing all the services programmed in, and reloading them with ADSIProg, or even before that trying voicemail, but the same thing always happens. 2. I can't seem to get call waiting id to work. I hear the adsi tones on the line when another call is coming in, but the phone doesn't seem to recognize it. Any ideas? 3. Is there a list or some documentation somewhere on what all the available adsi programming options there are, that I can use in .adsi files? Thanks. -Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
Hi Jim, This is an interesting product, especially for Cisco ATA-186 users..they can use one of the FXS ports to connect to the PSTN, but. you have a very limited functionality: when you call the phone number allocated to that specific port, you will get the tone for the PSTN line and can dial. When a call came from the PSTN, you will get an internal tone and can dial as a regular internal user. One idea is to allocate to that specific extension the prefix for the PSTN calls as extension number. Then you can call PSTN by dialing that prefix and then the number, with a sgort break. Overall. this is something that make sense to use for some specific purposes. Thanks for this info, Dan - - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 12:24 AM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? I think this is the company that makes them but it is hard to tell. http://www.artech.com.tw/html/english/AX300/AX300.htm This company sells them http://www.aislecom.com/ A rep. for them posted this thread, claimed to be the manufacturer. http://lists.digium.com/pipermail/asterisk-users/2003-March/009134.html There are quite a few comments so click on Next Message There is someone on eBay selling them http://tinyurl.com/bp4x disclaimer: I have never used one. I am not associated with the seller. On other lists I did hear some people had problems with them. You may want to start another thread and ask if any * users are using them. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 2:10 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? Hi, I look for something in the price range of a X100P for one FXO port. regarding the Dlink device, I think that there is not a real FXO port, more somethink like in Actiontec's InternetPhoneWizard, just to be able to use the analog phones for both IP and PSTN calls. It just switch one of the phone t the PSTN line. Have you considered a S100U and one of those $35 FXS to FXO converters? There is something like that? Where I can find such a converter and how this thing works? BR, Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 6:34 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk?
One more thing which can be a big issue with this device. It hangs the line ONLY based on busy tone... if not correctly detected, then it will keep the line open for ever, or you can select a call limit (15/30min.)/ Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 12:24 AM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? I think this is the company that makes them but it is hard to tell. http://www.artech.com.tw/html/english/AX300/AX300.htm This company sells them http://www.aislecom.com/ A rep. for them posted this thread, claimed to be the manufacturer. http://lists.digium.com/pipermail/asterisk-users/2003-March/009134.html There are quite a few comments so click on Next Message There is someone on eBay selling them http://tinyurl.com/bp4x disclaimer: I have never used one. I am not associated with the seller. On other lists I did hear some people had problems with them. You may want to start another thread and ask if any * users are using them. - Original Message - From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 2:10 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? Hi, I look for something in the price range of a X100P for one FXO port. regarding the Dlink device, I think that there is not a real FXO port, more somethink like in Actiontec's InternetPhoneWizard, just to be able to use the analog phones for both IP and PSTN calls. It just switch one of the phone t the PSTN line. Have you considered a S100U and one of those $35 FXS to FXO converters? There is something like that? Where I can find such a converter and how this thing works? BR, Dan - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 29, 2003 6:34 PM Subject: Re: [Asterisk-Users] External FXO device (USB or ethernet), supported by Asterisk? What price range are you looking for? Does anybody know if the FXO port of the Dlink DVG-1120 would work? http://www.dlink.com/products/voiceservices/dvg1120/ Have you considered a S100U and one of those $35 FXS to FXO converters? It is a nice thing overall, but I still need something much cheaper for home use. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] manager interface change request
hi all I'm trying to use the manager interface to do some nagios (http://nagios.org/) integration, and I find some parts of it not really optimal. What I'd like to change, is to make \r\n\r\n an actual terminator, something it isn't today, AFACS. Below is the Status output - it shows Response, Message, \r\n, Status post, \r\n, Status post etc etc. Without a parsable terminator, I need to use some select/poll interfaces, and I just don't like that :P May I suggest changing the \r\n between status (and other) output sections to something like '---\r\n'? regards roy action: status Response: Success Message: Channel status will follow Event: Status Channel: CAPI[contr2/22545070] CallerID: 22545070 State: Up Link: MGCP/aaln/[EMAIL PROTECTED] Event: Status Channel: MGCP/aaln/[EMAIL PROTECTED] CallerID: 22545070 State: Up Context: default Extension: 98013356 Priority: 1 Link: CAPI[contr2/22545070] -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] siemens optipoint 400 SIP
hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] External Directory Button and Dial tone on Cisco 7960 (SIP)
Has anyone found a solution how to use the directory button on the Cisco 7960? If configured correctly it should point to an external directory url. So far I failed to find any documentation regarding the format to set up a phone directory on my asterisk server. How can the dial tones on a CISCO 7960 be modified? Compared to the ATA 186, I could not find any settings that make a change possible. Thank you for your help Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAXTEL testing
Hi all, Just a quick one. Should I be able to call myself through IAXTEL using my 1700 number? I'm behind a NAT firewall and can call other numbers, I just want to test incoming calls somehow to make sure I can accept them from IAXTEL. Regards,Jamie Carl Email: [EMAIL PROTECTED] Phone: +61 414 365 466 Jabber: [EMAIL PROTECTED]
RE: [Asterisk-Users] manager interface change request
I concur! It would also help in parsing out the occasional junk I get on the socket. (I'm currently writing a wxwindows version of gastman) Also... I'm still not sure wheter I can be absolutely sure that the Responses will always be in the correct order... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: vrijdag 30 mei 2003 10:29 To: Asterisk mailing list Subject: [Asterisk-Users] manager interface change request hi all I'm trying to use the manager interface to do some nagios (http://nagios.org/) integration, and I find some parts of it not really optimal. What I'd like to change, is to make \r\n\r\n an actual terminator, something it isn't today, AFACS. Below is the Status output - it shows Response, Message, \r\n, Status post, \r\n, Status post etc etc. Without a parsable terminator, I need to use some select/poll interfaces, and I just don't like that :P May I suggest changing the \r\n between status (and other) output sections to something like '---\r\n'? regards roy action: status Response: Success Message: Channel status will follow Event: Status Channel: CAPI[contr2/22545070] CallerID: 22545070 State: Up Link: MGCP/aaln/[EMAIL PROTECTED] Event: Status Channel: MGCP/aaln/[EMAIL PROTECTED] CallerID: 22545070 State: Up Context: default Extension: 98013356 Priority: 1 Link: CAPI[contr2/22545070] -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] siemens optipoint 400 SIP
Thomas,
On Fri, 2003-05-30 at 08:22, Tomaz Izanc wrote:
hi!
anyone try siemens optipoint 400 economy SIP phone with * ?
Yes, it works pretty well and has message waiting indication.
and has rfc2833 if you apply a workaround.
But you need:
- Patch to replay contact address as is
* usually replies his own address in contact header
... this shall be corrected by Asterisk
all-chan_sip_030524.diff
- Patch that provides a workaround for message waiting
indication in rtp.c. This patch is just copied from a
previous workaround for Cisco.
This seems to be a problem of the OptiPoint 400.
... Siemens currently looks at the issue and
may has a solution soon.
rtp.c-op400-2833-workaround.diff
in sip.conf it looks like:
[3999]
type=friend
username=3999
host=dynamic
dtmfmode=rfc2833
callerid=3999 3999
mailbox=3999
context=SIPorig
BTWY:
What version has your Optipoint SIP-SW?
...mine is 1.1.4; I'm waiting for 2.0
you may call me at:
SIP:[EMAIL PROTECTED]
SIP:[EMAIL PROTECTED]
Willi
--- ./a-cvs/asterisk/channels/chan_sip.c2003-05-23 17:12:08.0 +0200
+++ ./a-wrk/asterisk/channels/chan_sip.c2003-05-24 13:41:21.0 +0200
@@ -1901,7 +1901,11 @@
snprintf(contact, sizeof(contact), %s;expires=%d, p-our_contact,
p-expiry);
snprintf(tmp, sizeof(tmp), %d, p-expiry);
add_header(resp, Expires, tmp);
+ /*ww lwc change header to copy
add_header(resp, Contact, contact);
+ */
+ copy_header(resp, req, Contact);
+
} else {
add_header(resp, Contact, p-our_contact);
}
@@ -2964,6 +2968,8 @@
char resp_hash[256];
char tmp[256] = ;
char *c;
+ /*ww need to check authorization headers for 'space' and ',' *z */
+ char *z;
char *response =;
char *resp_uri =;
@@ -3001,7 +3007,10 @@
}
} else
+ /*ww we need to check for spaces as well
c = strchr(c, ',');
+ */
+ if ((z = strchr(c,' ')) || (z = strchr(c,','))) c=z;
if (c)
c++;
}
--- ./a-cvs/asterisk/rtp.c 2003-05-16 04:50:46.0 +0200
+++ ./a-wrk/asterisk/rtp.c 2003-05-24 18:23:36.0 +0200
@@ -326,6 +326,12 @@
/* Comfort Noise */
f = process_rfc3389(rtp, rtp-rawdata + AST_FRIENDLY_OFFSET + hdrlen, res
- hdrlen);
if (f) return f; else return null_frame;
+ /* OP400 payloadtype 100 */
+ } else if (payloadtype == 100) {
+ /* Comfort Noise */
+ f = process_rfc2833(rtp, rtp-rawdata + AST_FRIENDLY_OFFSET + hdrlen, res
- hdrlen);
+ if (f) return f; else return null_frame;
+ /* end OP400 */
} else {
ast_log(LOG_NOTICE, Unknown RTP codec %d received\n, payloadtype);
return null_frame;
Re: [Asterisk-Users] chan_capi request
morning roy, yes, it's possible. the settings will move into the global section in 0.2.2. actually there is a use for a per-device gain configuration. you might like to have a capi device for outgoing calls to SCREAM at people (txgain=10) ... ;-) but i will add an option in the global section (so that the device setting overrides the global option). regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-05-28 um 14.09 schrieb Roy Sigurd Karlsbakk: hi all is it hard/possible to move the following from chan_capi_pvt.h into a setting (preferably global) in capi.conf? #define AST_CAPI_NATIONAL_PREF 0 #define AST_CAPI_INTERNAT_PREF 00 and ... Is it hard to move or copy the txgain and rxgain to [global], either as a given 'default' if nothing's set in the interfaces, or as a overall global. I just can't see the use of setting gain on individual interfaces -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] External Directory Button and Dial tone on Cisco 7960 (SIP)
Hi, How can the dial tones on a CISCO 7960 be modified? Compared to the ATA 186, I could not find any settings that make a change possible. Go to Settings SIP configuration 9 (Out of Band DTMF) You can choose between avt, avt_allways and none BR, Dan - Original Message - From: Ben Bosshardt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 30, 2003 12:26 PM Subject: [Asterisk-Users] External Directory Button and Dial tone on Cisco 7960 (SIP) Has anyone found a solution how to use the directory button on the Cisco 7960? If configured correctly it should point to an external directory url. So far I failed to find any documentation regarding the format to set up a phone directory on my asterisk server. How can the dial tones on a CISCO 7960 be modified? Compared to the ATA 186, I could not find any settings that make a change possible. Thank you for your help Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A Major Problem!
hi, we are experiecing the following probem, if anybody have come across such a problem or a solution to this please let us know. our set up is, an Asterisk server equipped with,4 port station interface card,single port fxo card and several soft sip phones we have found problems with the following scenarios, outside caller (calling through fxo interface) -- sip phone/ station interface phone calls to a conference outside caller (calling through fxo interface)--- confernce the problem is, once the outside caller(calling through fxo interface) disconnects the line, Asterisk does not detects the disconnection, other party can hear the 'engage like tone' coming from the other side.This continues till the other party(probalby the sip phone or the station interface phone) hangs up. If the fxo user was in a conference if he disconnets the line, other confencees can here the 'engage like tone' , this is very disturbing. The biggest problem is, the fxo line remains busy, till the sip/station phone user disconnects the line. Can anybody give us a solution for this. In thenear future, we are going to add some E1 lines too(with E400P cards), once this is done, will the above call disconnection problem occur in that configuration too..or is this a common problem only with analog ? Thank you very much, Surajee
