[Asterisk-Users] Asterisk with MySQL on Redhat 9
Hi I really hope somebody can help me out.
I have an asterisk installation working on a Redhat 9
system. I now want to add the MySQL functionally to it. However when I make the
necessary changes, (downloading the add-ons, and changing the Make file) the
make fails.
I have looked into this and I think I know what the problem
is. Basically I only have MySQL binaries installed. Can anyone advice me what
packages I need to install to get this going.
Help will be greatly appreciated.
Umar.
Registered in England No. 04348334. Tel: (+44) 0118 965 5600
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[Asterisk-Users] Delay Dial with Voicetronix
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had the
problem that chan_vpb.c is making a call before hearing the dialtone. The
suggestioin was to put a comma or more before the number and this would make
a pause before actually dialing the number. This seemed to be a probable
cause of my problems, so I've defined in extesnsions.conf:
[globals]
OUTDIAL=vpb/1-9/,,3487446196
[default]
exten = _55.,1,Dial(${OUTDIAL},30,r)
but this doesn't work, does someone have suggestions?
Tim
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Re: [Asterisk-Users] Asterisk with MySQL on Redhat 9
Umar Sear wrote: Hi I really hope somebody can help me out. I have an asterisk installation working on a Redhat 9 system. I now want to add the MySQL functionally to it. However when I make the necessary changes, (downloading the add-ons, and changing the Make file) the make fails. I have looked into this and I think I know what the problem is. Basically I only have MySQL binaries installed. Can anyone advice me what packages I need to install to get this going. Help will be greatly appreciated. Umar. You need to install the mysql and mysql-devel packages and any dependencies.. if you want the server to run on the same PC then you need to install mysql-server as well.. Later.. ___ 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] C++ and or C# .Net development contract for Asterisk PBX Management interface
Looking for a shining star in c# and or c++ .net development to take the reins on an Asterisk PBX management interface. The customer requests delivery of a working prototype by 9 April 2004, so time is of the essence. This is a paid contract and is open to a developer anywhere in the world. Expecting 40 - 80 hrs for contract fulfillment. All code will be released under the GPL back to the Asterisk community. The most important piece of the Asterisk package that does not yet exist - an easily configurable, Windows-based Management interface - could expand the Asterisk customer base to a whole new world of potential users. It should be understood that most receptionists who handle telephone calls aren't sitting in front of a Gastman-capable workstation. By developing this interface in a modular fashion, and opening it up to the open source community, we can all help to create the premier user interface for Asterisk. http://www.yottadot.com/callmanager/ There has already been a working prototype developed and will be an excellent reference for anyone not experienced in interfacing with the Asterisk Management API. The prototype can be found at the bottom of the page on the above link. If you're interested in this project, contact me via return email or utilizing one of the numbers listed below. If you know of someone who might be interested, please forward this information to them. Thank you for your time, Christian Hoffmeyer YottaDot Solutions Huntsville, AL (w) 256.859.4508 (c)256.655.0321 (iax) 700.859.4508 Ask me about Asterisk ___ 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 with MySQL on Redhat 9
In article [EMAIL PROTECTED], Umar Sear [EMAIL PROTECTED] wrote: I have an asterisk installation working on a Redhat 9 system. I now want to add the MySQL functionally to it. However when I make the necessary changes, (downloading the add-ons, and changing the Make file) the make fails. I have looked into this and I think I know what the problem is. Basically I only have MySQL binaries installed. Can anyone advice me what packages I need to install to get this going. You need the package mysql-devel In general if you want to compile a program that integrates some functionality of package X, you need to have package X-devel installed. 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
RE: [Asterisk-Users] local VoIP in Florida
772 is generally what cell phone companies in florida use. Nextel, Sprint, att, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Wednesday, March 17, 2004 10:21 PM To: Asterisk Users Subject: Re: [Asterisk-Users] local VoIP in Florida On Wed, Mar 17, 2004 at 08:54:57PM -0600, Matthew Marlowe wrote: 727 or 772? There is 772 in FL available. 727. That's St. Pete/Clearwater. What area is 772? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ 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] local VoIP in Florida
That's not actually correct. 772 area code is Port Saint Lucie, Fort Pierce, Vero Beach, Stuart, and Jensen Beach. That's Martin, Saint Lucie, and Indian River counties east coast of Florida just north of West Palm Beach. -Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Thursday, March 18, 2004 6:38 AM To: Asterisk Users Subject: RE: [Asterisk-Users] local VoIP in Florida 772 is generally what cell phone companies in florida use. Nextel, Sprint, att, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Wednesday, March 17, 2004 10:21 PM To: Asterisk Users Subject: Re: [Asterisk-Users] local VoIP in Florida On Wed, Mar 17, 2004 at 08:54:57PM -0600, Matthew Marlowe wrote: 727 or 772? There is 772 in FL available. 727. That's St. Pete/Clearwater. What area is 772? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 (888) 924-3728 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: line status
There are software solutions you could refer to, software based, or hardware based. Look at www.voip-info.org for both solutions available. To give you a hint; For Software based solution, you can try various managers available for asterisk (Windows and Linux based) in order to get line status. For hardware based, you could use an ADSI compliant phone, with big LCD screen, enough to show the channel info you need. If you can find it with soft-keys (interactively communicate with Asterisk through ADSI messages) that would be the greatest. Regards, Costas Halvajoglou Pre-Sales Engineer Fiber Systems Networks SA Greece -Original Message- From: Chris Clifton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 23 Feb 2004 23:26:35 -0500 Organization: Netlabz, Inc. Subject: [Asterisk-Users] line status Reply-To: [EMAIL PROTECTED] I've inquired about this before, but it seeems to me that most business class pbx systems allow the receptionist to see the status of all connected lines at a glance from their phone What are others doing to address this in the corporate environment ? If a receptionist has something like a Cisco 7960 + 7914 combination, sure, then he/she can transfer to any other extension via pre-programmed speed dial, but what about the status of the line ? Thanks, Chris Clifton ___ 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] Phantom problem authenticating with RSA?
I have three * servers that are inter-connected, registering with each other. Up until yesterday I was authenticating all three with MD5, and all was working fine. Yesterday I switched to RSA, and everything is working as well. I can see AUTHENTICATED messages on the console if one of the servers is restarted and reconnects, etc. Everything is working fine with calls being passed between them as well (which is why I labeled the subject Phantom problem). However, whenever a call is initiated between the servers I see the following NOTICE message: -- Called [EMAIL PROTECTED]/2001 -- Called [EMAIL PROTECTED]/2001 Mar 18 07:46:19 NOTICE[1150528304]: chan_iax2.c:3507 authenticate: No way to send secret to peer 'XX.XX.XX.XX' (their methods: 4) Mar 18 07:46:19 NOTICE[1150528304]: chan_iax2.c:3507 authenticate: No way to send secret to peer 'YY.YY.YY.YY' (their methods: 4) -- SIP/sipura-4b82 is ringing -- Call accepted by XX.XX.XX.XX (format ULAW) -- Format for call is ULAW -- IAX2[remote1]/3 stopped sounds -- Call accepted by YY.YY.YY.YY (format ULAW) Method 4 is RSA, which is what I have in all of the iax.conf files (below). The call shown above was successfully answered by a sipura device connected to remote2, so I am not having an authentication problem which is causing a problem at the user experience level, but this seems like something is still mis-configured on my part. Here are the iax.conf entires: on the local machine: [remote2] context=remote2-in type=friend host=remote2.com ; not the real name... auth=rsa inkeys=remote2 outkey=local [remote1] context=remote1-in type=friend host=remote1.com ; not the real name... auth=rsa inkeys=remote1 outkey=local on the remote1 machine: [remote2] context=remote2-in type=friend host=remote2.com auth=rsa inkeys=remote2 outkey=remote1 [local] context=local-in type=friend host=local.com auth=rsa inkeys=local outkey=remote1 on the remote2 machine: [local] context=from-local type=friend auth=rsa inkeys=local outkey=remote2 host=dynamic callgroup=1 pickupgroup=1 qualify=5 [remote1] context=from-local type=friend auth=rsa inkeys=remote1 outkey=remote2 host=dynamic callgroup=1 pickupgroup=1 qualify=5 Finally, since both local and remote1 are technically behind NAT firewalls, and remote2 is on a public IP address, I have register statements in both local and remote1 iax.conf files, and that's why the entries in remote2 have host=dynamic for those machines. I think that the qualify=5 statements are ignored in the iax.conf file, and I will remove them, but since they're in there now, I wanted to show the complete entries. Here are the register statements: on remote1: register = remote1:[EMAIL PROTECTED] on local: register = local:[EMAIL PROTECTED] Any help would be appreciated. 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
[Asterisk-Users] Softfax/spandsp
Hi all, It seems this week's release of spandsp fixed the major problems in the previous release, but still people have had a lot of trouble. Working with some of those who tried the software and gave me good feedback, I have identified some apparently common bugs in fax machines, and I have implemented workarounds for these in spandsp, and feedback so far seems good. I also fixed a couple of bugs. I think this version will work proper with a much wider range of fax machines. However, people have warning me that fax machines have a bad habit of not following the specs properly :-( There is now a new tarball at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try this, and report any problems you find. This version has the following changes: A floating point exception has been fixed A problem with the software not properly Some fax machines send a little less than the specified 1.5 seconds of training test data, so the training test failed every time. I now only look for 1.25seconds of training test data. Some fax machines do not correctly initialise the scrambler in their V.29 transmit modem. I have changed the software to tolerate this. Some fax machines send a burst of ones before the burst of zeros that forms the training test data. I have changed the software to tolerate this. 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] NuFone?
Haha, Well why not? Everyone has to eat at the end of the day! Is it worth considering setting up an asterisk-trading mailing list specifically for this purpose? Hotlinks Internet Services offers Voip grade bandwidth on our Juniper powered network and colocation space in the Major London datacenters including Telehouse London which is well known in the industry to have by far the widest choice of PSTN carriers available to connect to in the UK. We also offer call origination of 0207, 0845, 0870 and 0800 from the UK over IP or E1. Regards Panny Malialis [EMAIL PROTECTED] quakenet: panny-m00 icq: 36325362 msn: [EMAIL PROTECTED] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 17, 2004 8:17 PM Subject: RE: [Asterisk-Users] NuFone? Since everyone is offering their services then: USA - £0.016 (~ 2.9c) UK - £0.016 (~ 2.9c) Europe - £0.02 (~ 3.6c) UK 0800 - FREE SIP / IAX termination. auto-provisioning, web-based billing, call history, on-line top-up, credit-card payments. Not US-based though :-( Tan www.voiptalk.org www.iaxtalk.co.uk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer Sent: 17 March 2004 19:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NuFone? Doug Harris wrote: Hi, Seems like there arn't any alternative to NuFone either ? Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached. Doug If you want SIP/IAX termination from someone other than NuFone for the same price, you can contact me. We can offer that. 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 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] NuFone?
Well why not? Everyone has to eat at the end of the day! Is it worth considering setting up an asterisk-trading mailing list specifically for this purpose? But surely we'd all just end up trying to sell to each other that way! At least being on the main mailling list means that we have plenty of customers to prey on?! Hotlinks Internet Services offers Voip grade bandwidth on our Juniper powered ob: Magrathea offers A-Z IAX termination, origination blah blah blah blah. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NuFone?
Linus Surguy wrote: Well why not? Everyone has to eat at the end of the day! Is it worth considering setting up an asterisk-trading mailing list specifically for this purpose? But surely we'd all just end up trying to sell to each other that way! At least being on the main mailling list means that we have plenty of customers to prey on?! Bon Apetit Sir :) (If I misspelled above please correct me) ... ___ 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] * and PrePaid
There is a configuration and billing system for *. Refer to www.vidanetwork.com Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam
Re: [Asterisk-Users] NuFone?
Linus Surguy wrote: ob: Magrathea offers A-Z IAX termination, origination blah blah blah blah. I asked a while ago, and you passed me to a reseller who never answered my question - How much to terminate a call in the UK? David -- David Coulsonemail: [EMAIL PROTECTED] Linux Developer / web: http://davidcoulson.net/ Network Engineer phone: (216) 533-6967 ___ 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] thank u
From: ×áëâáôæüãëïõ Êþóôáò [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: line status Date: Thu, 18 Mar 2004 15:09:52 +0200 There are software solutions you could refer to, software based, or hardware based. Look at www.voip-info.org for both solutions available. To give you a hint; For Software based solution, you can try various managers available for asterisk (Windows and Linux based) in order to get line status. For hardware based, you could use an ADSI compliant phone, with big LCD screen, enough to show the channel info you need. If you can find it with soft-keys (interactively communicate with Asterisk through ADSI messages) that would be the greatest. Regards, Costas Halvajoglou Pre-Sales Engineer Fiber Systems Networks SA Greece -Original Message- From: Chris Clifton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 23 Feb 2004 23:26:35 -0500 Organization: Netlabz, Inc. Subject: [Asterisk-Users] line status Reply-To: [EMAIL PROTECTED] I've inquired about this before, but it seeems to me that most business class pbx systems allow the receptionist to see the status of all connected lines at a glance from their phone What are others doing to address this in the corporate environment ? If a receptionist has something like a Cisco 7960 + 7914 combination, sure, then he/she can transfer to any other extension via pre-programmed speed dial, but what about the status of the line ? Thanks, Chris Clifton ___ 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 _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk interoperability w/ new 64bit processors SIP express router
HEY! I'm doing research and testing for my Thesis on a prototype SIP PBX for a facility of 20-30 users. (T100P / Atlas 550series / Cisco Routers switches) A couple of concerns that have come up are: 1. Has there been any known issues concerning asterisk with the new 64-bit processors? 2. Asterisk is SIP compatible, but to my understanding it doesn't have support for SIP registrar, proxy or redirect server. Please correct me if wrong. I've yet to make a decision on which Sip server to use, so any ideas would be nice. SER was on my mind but the question is whether I can integrate it directly on the same linux server running Asterisk without complications or does it need to be separate. Comments, ideas and experiences would be greatly appreciated. These were a couple of subjects concerning me and couldn't seem to find answers to. By the way, I will be Documenting all testing and issues + much more on my homepages too. It will include a lot on SIP areas and ofcourse Asterisk* ! I'll make those available in the near future. Thanks ahead! t: Mike ___ 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_sccp latest cvs
How can I get the latest CVS of chan_sccp The way described on Zozos webpage seems not to work: [EMAIL PROTECTED]:~ export CVSROOT=:pserver:[EMAIL PROTECTED]:/var/lib/cvs/ [EMAIL PROTECTED]:~ cvs login (Logging in to [EMAIL PROTECTED]) CVS password: /var/lib/cvs/: no such repository cvs [login aborted]: authorization failed: server cvs.anlx.net rejected access [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] NuFone?
But surely we'd all just end up trying to sell to each other that way! At least being on the main mailling list means that we have plenty of customers to prey on?! It depends which way you look at it, it could just be more people to give you hassle and waste your time! I guess I was thinking more of carrier-carrier business. Panny ___ 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: Random Echo
I'm using some even older 32 ms 2551 and 2531's on my fxo and fxs lines. They work just like TC says. No training time, the echo is just gone. There is a serial, menu-driven interface on the Tellabs racks that makes them really easy to configure. My only complaint is that the rack is designed to hold 16 of them, is really big and needs a separate -48V supply. And the cabling on the newer 255 racks is optimized for 16 cans and is sort of complicated for only 1. I've started to build an enclosure for just 2 of them that runs from a 48v wall wart. I'm tired of explaining what the empty rack is for. Steve TC wrote: I did some google search but didn't find any details, about how to configure between Adtran 750 and T100P. If you have already done, please give us some details. not sure what level of dtl you want its quite straight fwd It varies depending on the chasis but in general there are T1 in and T1 out DB-15's for each T1 circuit you want to echo cancel. A straight T1 rj-45 cable goes to the channel bank other end is db-15 in to Tellab then a T1 X over goes from the Tellab db-15 out to the T100p card Then there are external Mode and Chan switchs that allow you to configure the T1 circuit (the line bld, framing, and coding/signaling), and then other setting to allow channels FXO/FXS LS, GS and enable echo cancel channel by channel, I do it for all fxs/fxo ports and turn ALL * echo cancel in zapata OFF.. ___ 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] Softfax/spandsp
Hi Steve. On Thu, 18 Mar 2004 22:06:46 +0800 Steve Underwood [EMAIL PROTECTED] wrote: snip There is now a new tarball at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try this, and report any problems you find. This version has the following changes: A floating point exception has been fixed A problem with the software not properly Some fax machines send a little less than the specified 1.5 seconds of training test data, so the training test failed every time. I now only look for 1.25seconds of training test data. Some fax machines do not correctly initialise the scrambler in their V.29 transmit modem. I have changed the software to tolerate this. Some fax machines send a burst of ones before the burst of zeros that forms the training test data. I have changed the software to tolerate this. Many thanks for your great release rxfax function is works well with my Canon MFC Multipass B-30!! mack_jpn ___ 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] Session numbers?
Hi, The messages produced by asterisk console, in vvv mode, what are the numbers after the brackets? in this example, /4 and /5 = Releasing [EMAIL PROTECTED]/4 and IAX2[ulf]/5 Are these session numbers or? Are they reused? When the first call comes after asterisk is restarted, they begin at /1 but 8 hours later, a new single call can have /4 I'm investigating why some calls do not go through to a Firefly client (IAX2) after the client has been busy. I'm suspecting som kind of zombie sessions... anyone? Any ideas? /Stig ___ 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] Fax termination in Asterisk
Hi everyone, Is there an application in Asterisk which can be used as a fax receiver? something like: exten = 1234,1,ReceiveFax(...) exten = 1234,2,ForwardReceivedFax( emailaddress ) Tomica
Re: [Asterisk-Users] Fax termination in Asterisk
Hi, You can build a solution with spandsp library. You will need an email server too. http://www.opencall.org/instruction Daniel Tomica Crnek wrote: Hi everyone, Is there an application in Asterisk which can be used as a fax receiver? something like: exten = 1234,1,ReceiveFax(...) exten = 1234,2,ForwardReceivedFax( emailaddress ) Tomica ___ 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] can't logon to voice mail - bad password
Search on DTMF - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 17, 2004 2:57 PM Subject: Re: [Asterisk-Users] can't logon to voice mail - bad password Paul, Do your other extensions work? If you have only one extension, note that the filename should be voicemail.conf ---^-- Just a thought ... Cheers, Willy - Original Message Follows - I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 = 3213,Bill Smith Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 Willy Wouters ypOne Publishing ___ 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] Re: Asterisk-Users] can't logon to voice mail - bad password
Paul, What Client are you using? Also what is the output on the console when you dial into the voicemail from that extension? I had the same issue using a BT100 set to early dial. It turned out to be a DTMF issue. Once I played with different DTMF options both on the phone and in the * configs, I managed to get it working. -Art - Original Message - From: Paul Mahler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, 2004-March-17 17:39 Subject: SPAM [Asterisk-Users] can't logon to voice mail - bad password I have one SIP extension that can't logon to voicemail. The log file says -- Incorrect password '3213' for user '4035' (context=other) even though the context in voicemail.cnf says 4035 = 3213,Bill Smith Thanks! Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 ___ 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] X-Lite on both sides of NAT with * behind the NAT
Hi, I'm confused about a config we have going where there is NAT router -- 192.168.1.101 linux+asterisk PC -- 192.168.1.104 WinXP with X-Lite At another location: NAT Router -- PC X-Lite xxx.xxx.xxx.xxx The remote works fine with *, can use the FXO line, can call FWD members thru *, can register with various services eveything seems to work fine. The problem I have now is that I want the person behind the NAT router at the office to be able to call me and transfer calls and vice-versa. Port 5060 and a bunch of ports starting at 1 are forwarded to the * box. How can I configure the PC at 192.168.1.104 to be able to talk both to * and thru * AND be handed off to me? Is it even possible? I tried setting the session port to 5063 and forwarded that to the PC at 104 and when I called the office we connected but there was no audio. Can someone clear up the mystery of the RTP ports for me? My home X-Lite starts at 8000 and I have a large number of ports above 8000 forwarded. I'm assuming the problem has to do with the RTP ports not being passed. thx, r ___ 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] Help configuring an Wildcard E100P
Hi !
I need a quick help configuring an Wildcard E100P ...
Inbound calls are working ok, but I can not call out, dialing 20 only
gets me a line dial tone, but no call is made; same stuff with _0.
direct dialing
Please provide some suggestions if you have ! TNX !
This is my actual config
zapata.conf
[channels]
signalling=pri_cpe
switchtype=euroisdn
group=1
context=default
channel = 1-15
channel = 17-31
zaptel.conf
loadzone = us
defaultzone = us
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
extentions.conf
[general]
static=yes
writeprotect=no
[globals]
[default]
exten = 20,1,Answer ; Answer the line
exten = 20,2,Dial(Zap/g1,0553024039)
exten = _0.,1,Dial,Zap/g1,${EXTEN:1}|45|r
exten = _0.,2,Congestion
--
Best regards,
Alessiomailto:[EMAIL PROTECTED]
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Re: [Asterisk-Users] * and PrePaid
It's just a short cut for Asterisk! In stead of spelling out the Asterisk PBX most just type *. - Original Message - From: Eric Kirkland To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 10:36 AM Subject: RE: [Asterisk-Users] * and PrePaid Ok, Im definitely too geeky. Im new to Asterisk, and Ive just started perusing through the lists, and I kept seeing messages referencing * in their conversations one, for example, was like Its amazing that there are still so few VoIP vendors that have support for *. Im like Wildcard? What, only a few vendors support fill in the blank?? Boy do I feel STPIT J Andy, [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Norman Sent: Thursday, March 18, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * and PrePaid There is a configuration and billing system for *. Refer to www.vidanetwork.com Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.576 / Virus Database: 365 - Release Date: 1/30/2004 ___ 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
RE: [Asterisk-Users] NuFone?
On Thu, 2004-03-18 at 07:47, Carey Jung wrote: Anybody have a list of area codes and prefixes for which Nufone can provide DIDs? I can't find any such list on their site. Michigan only, but I believe they have decent coverage within Michigan. I seem to recall they were planning on Chicago DIDs, but I don't know the status of that. ___ 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] Delay Dial with Voicetronix
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18 at 04:36, tim mickelson wrote:
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had the
problem that chan_vpb.c is making a call before hearing the dialtone. The
suggestioin was to put a comma or more before the number and this would make
a pause before actually dialing the number. This seemed to be a probable
cause of my problems, so I've defined in extesnsions.conf:
[globals]
OUTDIAL=vpb/1-9/,,3487446196
[default]
exten = _55.,1,Dial(${OUTDIAL},30,r)
but this doesn't work, does someone have suggestions?
Tim
_
Hitta rätt på nätet med MSN Sök http://search.msn.se/
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--
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
Unofficial Links section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/
BTEL Consulting
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[Asterisk-Users] Can i do voice chat without using the hardware
Hi, I am new to VOIP and Asterisk. I have downloaded and installed Asterisk in my Linux machine and tested using asterisk c command it works fine. It's an excellent product. Without using any of Digium's hardware or T1 or E1 interfaces , can i do voice chat between two computers (intranet/internet)? If possible, How can i do that? (Any configuaration setting is required?) Waitng for your help. Regards, Sur __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] openh323 w/t38
Hello, I have H323 up and running. However, I do not see the T38 codec as an option. I have looked through the mailing-list and saw a couple of postings with T38 listed in the codec list for the oh323.conf file. Am I missing something here? Regards, Mark
[Asterisk-Users] h323 Dialing newbie Question?
I am using NuFone H323 module. Following on extensions.conf works (x.x.x.x = is the IP address) extensions.conf --- exten = 2000,1,Dial(H323/[EMAIL PROTECTED]) Following do not seems to work, but I need to dial out using following, due to various reasons. Why I cannot dial out using following format. extensions.conf --- exten = 2000,1,Dial(H323/[EMAIL PROTECTED]) h323.conf - [h323-dial] type=peer host=x.x.x.x In addition I tried type=peer type=h323 etc nothing seems to work. Documentation is not very helpful and there is nothing I can find on the Message Board archive. Any help/hints appreciated. - SamW ___ 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] Problems with asterisk and gnophone on Gentoo box
On 18/03/04 15:40, Kevin wrote: I seem to be having problems using my sound card with asterisk and gnophone in a Gentoo system (not sure if it being Gentoo is important or not, but thought I'd mention it just in case). I have the following errors when starting gnophone: Looks to me like you're probably using ALSA but you don't have its OSS compatibility layer enabled. emerge alsa-oss Check out: - http://www.gentoo.org/doc/en/alsa-guide.xml Regards, Alastair ___ 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] Softfax/spandsp
Hi, -Original Message- Some fax machines send a little less than the specified 1.5 seconds of training test data, so the training test failed every time. I now only look for 1.25seconds of training test data. I think this is still on the long side? I have a few fax-services that seem to be hard to handshake with. Below is a sample. By the way, it may be a little cluttered because another fax was coming in slightly earlier. Perhaps an idea to start debug logging with the channel its running on ;-) Florian -- Executing RxFAX(CAPI[contr1/534280109]/7, /var/spool/asterisk/fax/fax20040318-173618.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down Slow carrier up TSI: 43 37 39 35 31 35 34 35 33 32 31 33 2b 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: +31235451597 DCS: 83 00 06 70 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1653.68 (17) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1699.60 (403) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1699.33 (86) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 1 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.04 (401) Fast carrier down Fast carrier up Coarse carrier frequency 1698.87 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.05 (2975) Fast carrier down Fast carrier up Coarse carrier frequency 1700.08 (2975) Fast carrier down Fast carrier up Coarse carrier frequency 1699.76 (2988) Fast carrier down Fast carrier up Coarse carrier frequency 1699.92 (2984) Fast carrier down Fast carrier up Coarse carrier frequency 1209.55 (3) Fast carrier down cdr_odbc: Query Successful! -- CAPI Hangingup ___ 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] Monastery Devel snapshot
I just pushed out a snapshot of the -devel version of monastery. ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] MWI only working after handset was lifted once
Hello, All. I have a bit of a peculiar problem with MWI. First some basics: Hardware: PT390 connected to a Digium TDM400P I included mailbox=100 statement in zapdata.conf and the mailbox is defined in voicemail.conf. Everything works OK. I receive VMs, and when I pick up the handset I get the stutter tone. However, the MWI light on the PT390 ** does not come on until the handset was picked up once***. After lifting the handset and returning it to the hook it comes on and indicates correctly if there is a VM. or not. Am I missing something? Oliver ___ 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] Should List be Moderated?
This was posted last year by Mark. I figured I'd repost it to refresh peoples memories. Please stop posting commercial postings and announcements to the *-users and *-dev. Let's self moderate so the list doesn't have to be moderated James -- Forwarded message -- Date: Mon, 7 Apr 2003 18:43:17 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Should List be Moderated? In light of recent flame baits and advertisements sent to the list, I would like to seek opinions of list members on making the list moderated. I certainly don't have time to moderate the list myself, so I would suggest giving at least a half dozen, maybe more, people the ability to approve posts to keep it flowing quickly. Moderators would be asked just to approve/disapprove based upon a specific list of characteristics. Among characteristics that *could* be considered: * Posts should not advertise products, especially not those unusuable under Asterisk * Posts should not contain profanity * Posts should not simply be me-too's * Arguably, maybe something related to flame baits Any comments on any of these rules, or suggestions for others, that would make the list more valuable? Mark ___ 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Can i do voice chat without using the hardware
Sure thing. You're going to have to get SIP involved though. This means using sip.conf to create new sip users. Do a search on www.voip-info.org for sip.conf and it will explain how to configure a user for SIP. Then you'll need SIP clients (hard VoIP phones, or SIP soft clients such as Windows Messenger or X-Lite). You can make VoIP calls over an existing network infrastructure without analog hardware. For instance, I have an internal Asterisk PBX allowing VoIP conversations between X-Lite, Windows Messenger, and Pingtel clients - all over networking connections, no T1/E1/Analog needed. You need the hardware when you start interfacing with the PSTN for the most part. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of suresh kumar Sent: March 18, 2004 11:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Can i do voice chat without using the hardware Hi, I am new to VOIP and Asterisk. I have downloaded and installed Asterisk in my Linux machine and tested using asterisk -c command it works fine. It's an excellent product. Without using any of Digium's hardware or T1 or E1 interfaces , can i do voice chat between two computers (intranet/internet)? If possible, How can i do that? (Any configuaration setting is required?) Waitng for your help. Regards, Sur __ Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam http://mail.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Pulver WiSIP Dual Line and Hold?
I don't think it can do these things. Yes I know the web pages says so but the book doesn't and neither does Yan, Pulvers techy. Mark Steven Thomas said: Hi, I have received my WiSIP phone - works well for basic functions of call answer and hang-up! Does anyone know how to enable Dual line support, Hold and Transfer functions with this phone via Asterisk. Thanks, Regards, Steven Thomas -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Should List be Moderated?
So give us a commercial list. Please :) Panny - Original Message - From: James Golovich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 5:10 PM Subject: [Asterisk-Users] Should List be Moderated? This was posted last year by Mark. I figured I'd repost it to refresh peoples memories. Please stop posting commercial postings and announcements to the *-users and *-dev. Let's self moderate so the list doesn't have to be moderated James -- Forwarded message -- Date: Mon, 7 Apr 2003 18:43:17 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Should List be Moderated? In light of recent flame baits and advertisements sent to the list, I would like to seek opinions of list members on making the list moderated. I certainly don't have time to moderate the list myself, so I would suggest giving at least a half dozen, maybe more, people the ability to approve posts to keep it flowing quickly. Moderators would be asked just to approve/disapprove based upon a specific list of characteristics. Among characteristics that *could* be considered: * Posts should not advertise products, especially not those unusuable under Asterisk * Posts should not contain profanity * Posts should not simply be me-too's * Arguably, maybe something related to flame baits Any comments on any of these rules, or suggestions for others, that would make the list more valuable? Mark ___ 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 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] Several H323 bugfixes - working SIP - H.323 translator
I just tried this, and it's not working for me.. I can't call a 2600 or a
CCM... What version of OpenH323 and PWLIB did you all use?
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
--
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
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[Asterisk-Users] SIP problem with Nikotel
Hi, I'm testing Nikotel with Asterisk. Sound quality is Ok, but I can´t manage to have a call longer then 1 minute After 1 minute or so, my * exchanges some SIP messages with Nikotel and the call ends with maximum retries error. Debugging the SIP messages, I see 2 IP´s in the VIA header, the calamar0.nikotel.com (63.214.186.6 Nikotel server) and a CiscoSystemsSIP (195.126.99.75). I suppose this second IP is part of the Nikotel routers. The problem is that, when my Asterisk sends a INVITE message including the second IP, the same Cisco returns a 606 message. After a couple of this messages, the server hungs up with maximum retries. Following are the messages I'm talking about. Any ideas? Am I doing something wrong? Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 80.28.41.108:6012;branch=z9hG4bK65f066d4 From: 13101 sip:[EMAIL PROTECTED]:5082;tag=as14c97a22 To: sip:[EMAIL PROTECTED];tag=18B7B6CC-1BEC Date: Thu, 18 Mar 2004 14:34:34 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Allow-Events: telephone-event Content-Disposition: session;handling=required Content-Type: application/sdp Content-Length: 286 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:[EMAIL PROTECTED]:5060;maddr=63.214.186.6 v=0 o=CiscoSystemsSIP-GW-UserAgent 4092 6217 IN IP4 195.126.99.75 s=SIP Call c=IN IP4 195.126.99.75 t=0 0 m=audio 16842 RTP/AVP 3 101 100 c=IN IP4 195.126.99.75 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 14 headers, 12 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format telephone-event Found description format X-NSE Capabilities: us - 1550, them - 2/0, combined - 2 Non-codec capabilities: us - 1, them - 1, combined - 1 -- SIP/nikotel-c3c9 is making progress passing it to [EMAIL PROTECTED]/10 bcn01*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.28.41.108:6012;branch=z9hG4bK65f066d4 From: 13101 sip:[EMAIL PROTECTED]:5082;tag=as14c97a22 To: sip:[EMAIL PROTECTED];tag=18B7B6CC-1BEC Date: Thu, 18 Mar 2004 14:34:34 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Session-Expires: 120;refresher=uas Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 286 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:[EMAIL PROTECTED]:5060;maddr=63.214.186.6 v=0 o=CiscoSystemsSIP-GW-UserAgent 4092 6217 IN IP4 195.126.99.75 s=SIP Call c=IN IP4 195.126.99.75 t=0 0 m=audio 16842 RTP/AVP 3 101 100 c=IN IP4 195.126.99.75 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 15 headers, 12 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format telephone-event Found description format X-NSE Capabilities: us - 1550, them - 2/0, combined - 2 Non-codec capabilities: us - 1, them - 1, combined - 1 list_route: hop: sip:[EMAIL PROTECTED]:5060;maddr=63.214.186.6 list_route: hop: sip:[EMAIL PROTECTED]:5060 set_destination: Parsing sip:[EMAIL PROTECTED]:5060;maddr=63.214.186.6 for address/port to send to -- SIP/nikotel-c3c9 answered [EMAIL PROTECTED]/10 set_destination: set destination to 63.214.186.6, port 5060 Transmitting: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 80.28.41.108:5082;branch=z9hG4bK65f066d4 Route: sip:[EMAIL PROTECTED]:5060 From: 13101 sip:[EMAIL PROTECTED]:5082;tag=as14c97a22 To: sip:[EMAIL PROTECTED];tag=18B7B6CC-1BEC Contact: sip:[EMAIL PROTECTED]:5082 Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 63.214.186.6:5060 bcn01*CLI Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 80.28.41.108:6012;branch=z9hG4bK65f066d4 From: 13101 sip:[EMAIL PROTECTED]:5082;tag=as14c97a22 To: sip:[EMAIL PROTECTED];tag=18B7B6CC-1BEC Date: Thu, 18 Mar 2004 14:34:34 GMT Call-ID: [EMAIL PROTECTED] Server: Cisco-SIPGateway/IOS-12.x CSeq: 103 INVITE Session-Expires: 120;refresher=uas Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 286 Contact: sip:[EMAIL PROTECTED]:5060 Record-Route: sip:[EMAIL PROTECTED]:5060;maddr=63.214.186.6 v=0 o=CiscoSystemsSIP-GW-UserAgent 4092 6217 IN IP4 195.126.99.75 s=SIP Call c=IN IP4 195.126.99.75 t=0 0 m=audio 16842 RTP/AVP 3 101 100 c=IN IP4 195.126.99.75 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 15 headers, 12 lines Found audio format UNKN Found audio format UNKN Found audio format UNKN Found description format GSM Found description format telephone-event Found description format X-NSE Capabilities: us - 1550, them - 2/0, combined - 2 Non-codec capabilities: us - 1, them - 1, combined - 1
Re: [Asterisk-Users] Delay Dial with Voicetronix
It is from this extensive thread that I fond that I should put a comma in the
dial string, that didn't help, now what should I do? This thread regarding this
issue does not help me.
tim
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18 at 04:36, tim mickelson wrote:
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had the
problem that chan_vpb.c is making a call before hearing the dialtone. The
suggestioin was to put a comma or more before the number and this would make
a pause before actually dialing the number. This seemed to be a probable
cause of my problems, so I've defined in extesnsions.conf:
[globals]
OUTDIAL=vpb/1-9/,,3487446196
[default]
exten = _55.,1,Dial(${OUTDIAL},30,r)
but this doesn't work, does someone have suggestions?
Tim
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Re: [Asterisk-Users] Should List be Moderated?
And thus Asterisk-Biz was born. (http://lists.digium.com/mailman/listinfo/asterisk-biz) On Thu, 18 Mar 2004, Panny Malialis wrote: So give us a commercial list. Please :) Panny - Original Message - From: James Golovich [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 5:10 PM Subject: [Asterisk-Users] Should List be Moderated? This was posted last year by Mark. I figured I'd repost it to refresh peoples memories. Please stop posting commercial postings and announcements to the *-users and *-dev. Let's self moderate so the list doesn't have to be moderated James -- Forwarded message -- Date: Mon, 7 Apr 2003 18:43:17 -0500 (CDT) From: Mark Spencer [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Should List be Moderated? In light of recent flame baits and advertisements sent to the list, I would like to seek opinions of list members on making the list moderated. I certainly don't have time to moderate the list myself, so I would suggest giving at least a half dozen, maybe more, people the ability to approve posts to keep it flowing quickly. Moderators would be asked just to approve/disapprove based upon a specific list of characteristics. Among characteristics that *could* be considered: * Posts should not advertise products, especially not those unusuable under Asterisk * Posts should not contain profanity * Posts should not simply be me-too's * Arguably, maybe something related to flame baits Any comments on any of these rules, or suggestions for others, that would make the list more valuable? Mark ___ 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 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] Monastery Devel snapshot
... just installed this. The database updates OK but status.php shows no active channels (either SIP to SIP or SIP to voicemail). Iain --On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED] wrote: I just pushed out a snapshot of the -devel version of monastery. ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] Several H323 bugfixes - working SIP - H.323 translator
Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. Did you put these files to bugs.digium.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] Asterix Sip Stack
Hi, Could you tell me what role the ASTERIX can play. Is it Sip Registry Server ?. Could it work as Proxy Server ? Thanks Ahmet BerliKomm Telekommunikationsgesellschaft mbH Ahmet Balamir Phone:+49 30 8188 9821 Ludwig-Erhard-HausFax: Fasanenstraße 85CellPhone: +49 163 818 9821 10623 Berlin eMail:[EMAIL PROTECTED] Germany WWW:http://www.berlikomm.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
[Asterisk-Users] Re: Random Echo
TC, Appreciated your help and will try out TelLabs card and see if we can get rid of echo. Yesterday I did some changes in TX and RX attenuation setting on Channel bank and it reduces the echo, but it is not yet vanished as we wanted. Any way Thanks. Regards, KD Date: Wed, 17 Mar 2004 19:47:16 -0800 From: TC [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Random Echo To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I did some google search but didn't find any details, about how to configure between Adtran 750 and T100P. If you have already done, please give us some details. not sure what level of dtl you want its quite straight fwd It varies depending on the chasis but in general there are T1 in and T1 out DB-15's for each T1 circuit you want to echo cancel. A straight T1 rj-45 cable goes to the channel bank other end is db-15 in to Tellab then a T1 X over goes from the Tellab db-15 out to the T100p card Then there are external Mode and Chan switchs that allow you to configure the T1 circuit (the line bld, framing, and coding/signaling), and then other setting to allow channels FXO/FXS LS, GS and enable echo cancel channel by channel, I do it for all fxs/fxo ports and turn ALL * echo cancel in zapata OFF.. ___ 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] Softfax/spandsp
Title: RE: [Asterisk-Users] Softfax/spandsp Hi, This week's spandsp release is a big step ahead, there are no problems so far receiving faxes from our HP OfficeJet R80xi and Panafax UF-560. Still, we cannot get anything from our Dialogic fax boards. It goes like this: -- Executing RxFAX(Zap/42-1, /usr/tmp/nativefax.tif) in new stack Changed from phase 0 to 1 Slow carrier up [03/18/04 11:48:05.886] DEBUG[442394]: File chan_zap.c, Line 3332 (zt_exception): Exception on 57, channel 42 [03/18/04 11:48:05.886] DEBUG[442394]: File chan_zap.c, Line 2753 (zt_handle_event): Got event No event(0) on channe l 42 (index 0) [03/18/04 11:48:05.886] DEBUG[442394]: File chan_zap.c, Line 3234 (zt_handle_event): Dunno what to do with event 0 o n channel 42 Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 4e 49 42 55 52 41 5a 20 58 45 4c 41 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: ALEX ZARUBIN DCS: 83 00 c6 70 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1656.15 (8) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1701.02 (74) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1700.01 (4854) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1701.27 (74) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1700.08 (4871) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1701.12 (74) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1699.79 (2940) Fast carrier down 0 bad bits in trainability test FTT: 44 ^M -- Channel 18, span 2 got hangup [03/18/04 11:48:35.668] DEBUG[442394]: File app_rxfax.c, Line 200 (rxfax_exec): Got hangup Thank you. Alex Zarubin Webley Systems -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED]] Sent: Thursday, March 18, 2004 8:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Softfax/spandsp Hi all, It seems this week's release of spandsp fixed the major problems in the previous release, but still people have had a lot of trouble...
Re: [Asterisk-Users] Should List be Moderated?
Now if we can just get the list software configured to bounce untrimmed posts with multiple copies of the footer, we would be all set!! ___ 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 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 Oh yeah, for the humor impaired: :-) ___ 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] Should List be Moderated?
quote who=James Golovich And thus Asterisk-Biz was born. (http://lists.digium.com/mailman/listinfo/asterisk-biz) [EMAIL PROTECTED]: unknown user: asterisk-biz-request So, when will it be fully up? -- END OF LINE -MCP ___ 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] Several H323 bugfixes - working SIP - H.323 translator
Will these be available on the CVS? Devel or Stable? Hi all, in an effort to create a SIP - H.323 translator we've found and fixed several problems in H.323 channel. These inlcude: for SIP-H.323 calls - no ringback tone - ringback not related to H.323 events - one-way audio with Cisco CallManager - incorrect Caller ID for H.323-SIP calls - not able to establish call with Cisco IOS 12.3(4)T - ringback not related to SIP events - no support for 183 Call Progress - incorrect Caller ID Please find the patches against aterisk 0.7.2 release below. M. ___ 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] Problems with asterisk and gnophone on Gentoo box
On Thursday 18 March 2004 11:29, Alastair Maw wrote: On 18/03/04 15:40, Kevin wrote: I seem to be having problems using my sound card with asterisk and gnophone in a Gentoo system (not sure if it being Gentoo is important or not, but thought I'd mention it just in case). I have the following errors when starting gnophone: Looks to me like you're probably using ALSA but you don't have its OSS compatibility layer enabled. emerge alsa-oss Check out: - http://www.gentoo.org/doc/en/alsa-guide.xml Thanks for your reply, Alastair. I did use that guide in getting myself set-up with sound, and do have alsa-oss installed: bash-2.05b# epm -qa | grep alsa alsa-lib-1.0.2 alsa-oss-1.0.2 alsa-utils-1.0.2 alsa-tools-1.0.2 alsaplayer-0.99.75-r1 alsa-driver-1.0.2c Is there any other special configuration of asterisk that needs to be done? I know I made some manual changes to the /etc/modules.d/alsa file during my initial setup of sound. Does asterisk need a special alias in it perhaps, or should my /dev/dsp be an alias to something else (not that I understand these aliases all that well...)? bash-2.05b# cat /etc/modules.d/alsa| grep -v \# alias char-major-116 snd alias char-major-14 soundcore alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss options snd cards_limit=1 bash-2.05b# Thanks again for your reply. -Kevin ___ 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] Any ISDN BRI card recommendations for North America?
Rob, Thanks for the info. Since it seems like BRI is not too popular in the U.S., I think that I will try to pick up a DIVA PCI and see if it will work with CAPI or i4l. -Tor Roberts Rob Fugina wrote: On Wed, Mar 17, 2004 at 12:25:49PM -0800, Tor Roberts wrote: Hi all, I have been using Asterisk for a couple of months now with some GS handsets and an X100P FXO card. The system works great, but I would like to add ISDN BRI to take advantage of the extra features, faster call setup time, etc. I was wondering if anyone could recommend any BRI cards that work in the U.S. and don't cost a fortune. I have checked the archieves and it seems like there are not many people in the U.S. using BRI. I was hoping that I could use either an Eicon DIVA PCI or an Eicon DIVA PRO as they are not that too expensive. If anyone has used either of these cards in the U.S., or can recommend another alternative, that would be great! I'm looking for the same thing, for the same reasons. I believe the PRO version is not supported by the i4l drivers. Someone referred me to the following product, made by an Australian company it seems, but I haven't been able to find a source... To be honext, I haven't tried too hard -- busy with other things. http://www.traverse.com.au/productview.do?product_id=14 Rob ___ 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] NuFone?
Is there any reason the only country you can choose is USA? There are more countries than that... :-) Matt I thought thats what http://www.iaxtel.org was all about.. :) ___ 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] Delay Dial with Voicetronix
On 2004 Mar 18, at 11:29, tim_mickelson wrote: It is from this extensive thread that I fond that I should put a comma in the dial string, that didn't help, now what should I do? This thread regarding this issue does not help me. You cannot put either the , or the directly into a Dial string. I had a patch on the bugtracker to solve this, but apparently it does not work (still haven't had time to play with the VoiceTronix card I have). The workaround is to get a T100P and a channel bank with FXO ports. -Tilghman ___ 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] Delay Dial with Voicetronix
From my extensions.conf:
[globals]
#TRUNK=Zap/1
VPBPAUSE=,
[trunkld]
exten = _91NX,2,Dial(vpb/1-1/${VPBPAUSE}${EXTEN:1})
exten = _91NX,1,Dial(vpb/1-2/${VPBPAUSE}${EXTEN:1})
On Thu, 2004-03-18 at 11:29, tim_mickelson wrote:
It is from this extensive thread that I fond that I should put a comma in the
dial string, that didn't help, now what should I do? This thread regarding this
issue does not help me.
tim
Check the extensive thread regarding this EXACT ISSUE in the mailing
list archives.
On Thu, 2004-03-18 at 04:36, tim mickelson wrote:
Hi.
I'm not being able to make my Voicetronix Openswitch 12 work with
Asterisk. When making a call from a SIP-phone to the PSTN, the SIP-phone is
ringing, the Asterisk says that it is ringing, but the phone I'm ringing is
not ringing. I've seen in the mail list that other people have had the
problem that chan_vpb.c is making a call before hearing the dialtone. The
suggestioin was to put a comma or more before the number and this would make
a pause before actually dialing the number. This seemed to be a probable
cause of my problems, so I've defined in extesnsions.conf:
[globals]
OUTDIAL=vpb/1-9/,,3487446196
[default]
exten = _55.,1,Dial(${OUTDIAL},30,r)
but this doesn't work, does someone have suggestions?
Tim
_
Hitta rätt på nätet med MSN Sök http://search.msn.se/
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For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
Unofficial Links section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/
BTEL Consulting
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Re: [Asterisk-Users] NuFone?
Eric Wieling wrote: Michigan only, but I believe they have decent coverage within Michigan. I seem to recall they were planning on Chicago DIDs, but I don't know the status of that. Jeremy from Nufone and Mark Spencer were both at this week's WISPCON in Chicago. From the smell of it we won't have to wait very long before there will be a number of choices wrt nationwide DID that then can interconnect via a number of VoIP protocols, including IAX. There were several CLEC types there who (at least from their telling) have pretty big footprints, and it sure seems that asterisk is catching everyone's attention. It is unfortunate but understandable that as these ITSPs begin to roll out their services, there are going to growing pains. There is really IMO no existing business model into which they cleanly fit, and from the perspective of customer service, most of them are growing too fast. Lot of those WISPs have been playing with lots of ITSPs, and the sense of the house was that it is a rare one that doesn't have occasional-to-frequent termination problems. Nufone and Vonage were basically the only ones there for which those complaints weren't raised. And, FWIW, I have not had hugely good luck with Voicepulse in terms of responsiveness of their customer service effort. B. ___ 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] Schools/Districts using asterisk?
I'm investigating asterisk to use as a replacement for an aging Lucent PBX in our district office, as well as replacing the Centrex/intercom based systems at our schools. I'm curious if any other schools/districts are using asterisk? If so, I'd certainly be interested in talking about your setup. Thanks, -- Chris Hobbs Silver Valley Unified School District Head geek: Technology Services Coordinator webmaster: http://www.silvervalley.k12.ca.us/~chobbs/ postmaster: [EMAIL PROTECTED] pgp: http://www.silvervalley.k12.ca.us/~chobbs/key.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, X100P and ATT PBX
Yesterday I tried to connect an * server with an X100P card to an extension of an ATT PBX. The X100P never could detect the line and always gave an alarm. Is there some special type of config that must be done to connect an FXO port to an extension of a PBX? -- Carlos Chavez Corporativo Lacer S.A. de C.V. ___ 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] Problems with asterisk and gnophone on Gentoo box
On Thursday 18 March 2004 13:46, Kevin wrote: On Thursday 18 March 2004 11:29, Alastair Maw wrote: On 18/03/04 15:40, Kevin wrote: I seem to be having problems using my sound card with asterisk and gnophone in a Gentoo system (not sure if it being Gentoo is important or not, but thought I'd mention it just in case). I have the following errors when starting gnophone: Looks to me like you're probably using ALSA but you don't have its OSS compatibility layer enabled. emerge alsa-oss Check out: - http://www.gentoo.org/doc/en/alsa-guide.xml Thanks for your reply, Alastair. I did use that guide in getting Just noticed that my grep/paste from /etc/modules.d/alsa (the contents of this file end up in /etc/modules.conf after running a script) left out some active lines with comments near the end. Correct /etc/modules.d/alsa follows: bash-2.05b$ cat /etc/modules.d/alsa|grep -v ^\# alias char-major-116 snd alias char-major-14 soundcore alias snd-card-0 nvaudio # testing alias sound-slot-0 snd-card-0 # testing alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss options snd cards_limit=1 bash-2.05b$ Also noticed some other messages that don't come up in asterisk with every start: * [chan_oss.so] = (OSS Console Channel Driver) Mar 18 14:29:57 WARNING[16384]: chan_oss.c:352 setformat: Requested 8000 Hz, got 7866 Hz -- sound may be choppy Mar 18 14:29:57 WARNING[16384]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found [app_db.so] = (Database access functions for Asterisk extension logic) == Registered application 'DBget' == Registered application 'DBput' == Registered application 'DBdel' == Registered application 'DBdeltree' [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Mar 18 14:29:57 WARNING[229391]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI dial *CLI -- Executing Wait(OSS/dsp, 1) in new stack -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing DigitTimeout(OSS/dsp, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(OSS/dsp, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(OSS/dsp, demo-congrats) in new stack -- Playing 'demo-congrats' (language 'en') Mar 18 14:33:59 WARNING[262161]: chan_oss.c:408 soundcard_setinput: Unable to re-open DSP device: Device or resource busy Mar 18 14:33:59 WARNING[262161]: chan_oss.c:567 oss_write: Unable to set device to input mode Mar 18 14:33:59 WARNING[262161]: file.c:521 ast_readaudio_callback: Failed to write frame == Spawn extension (local, s, 5) exited non-zero on 'OSS/dsp' Hangup on console *CLI * That seems to be an indicator that this onboard sound-card is not full duplex capable, but I'm skeptical of that conclusion simply because I think it's pretty high-end. Also because I can operate artsd in full duplex mode without apparent problems. Any thoughts? TIA. -Kevin ___ 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, X100P and ATT PBX
Carlos Chavez wrote: Yesterday I tried to connect an * server with an X100P card to an extension of an ATT PBX. The X100P never could detect the line and always gave an alarm. Is there some special type of config that must be done to connect an FXO port to an extension of a PBX? I had that exact same problem, and it turned out to be the wiring of the jumper I was using to the FXO port had the two leads reversed. Maybe worth a try? B. ___ 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] Schools/Districts using asterisk?
We are using Asterisk in K12 with similar goals. We have pretty much decided to go with it district wide. It will be a 200-400 phone installation across 9 sites. We also looking at eliminating a Centrex system (called Plexar in our area). Current status is that I have patched the * in front of our existing legacy system. This allowed testing of new phones while be got comfortable with * being stable, etc. Now I am just shopping for phones before we pull out the legacy system and start the process of moving schools off of the Centrax contract. Something you should look at in a school setting is the Wisip wireless 802.11b phone from Pulver. We are looking at this to help reduce cellular costs and also as a possible teacher phone. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Chris Hobbs [EMAIL PROTECTED]: I'm investigating asterisk to use as a replacement for an aging Lucent PBX in our district office, as well as replacing the Centrex/intercom based systems at our schools. I'm curious if any other schools/districts are using asterisk? If so, I'd certainly be interested in talking about your setup. Thanks, -- Chris Hobbs Silver Valley Unified School District Head geek: Technology Services Coordinator webmaster: http://www.silvervalley.k12.ca.us/~chobbs/ postmaster: [EMAIL PROTECTED] pgp: http://www.silvervalley.k12.ca.us/~chobbs/key.asc ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] Asterix Sip Stack
Hello Ahmet, Asterisk is more than a proxy. Its an entire PBX. At a basic level it can be used as a proxy though. [EMAIL PROTECTED] said: Hi, Could you tell me what role the ASTERIX can play. Is it Sip Registry Server ?. Could it work as Proxy Server ? Thanks Ahmet BerliKomm Telekommunikationsgesellschaft mbH Ahmet Balamir Phone:+49 30 8188 9821 Ludwig-Erhard-HausFax: Fasanenstraße 85CellPhone: +49 163 818 9821 10623 Berlin eMail: [EMAIL PROTECTED] Germany WWW:http://www.berlikomm.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 -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Problems with asterisk and gnophone on Gentoo box
On 18/03/04 18:46, Kevin wrote: Thanks for your reply, Alastair. I did use that guide in getting myself set-up with sound, and do have alsa-oss installed: You need to have it all insmod'ed as well (which I guess it will be): [EMAIL PROTECTED] almaw # lsmod | grep oss snd-seq-oss29216 0 snd-seq-midi-event 3584 0 [snd-seq-oss] snd-seq37584 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4304 0 [snd-rawmidi snd-seq-oss snd-seq] snd-pcm-oss38436 0 snd-pcm60960 0 [snd-via82xx snd-pcm-oss] snd-mixer-oss 13680 0 [snd-pcm-oss] snd33636 1 [...snip...] Also make sure your dsp device is accessible for the user running OSS: [EMAIL PROTECTED] almaw # ls -l /dev/dsp lr-xr-xr-x 1 root root 9 Mar 9 10:02 /dev/dsp - sound/dsp [EMAIL PROTECTED] almaw # ls -l /dev/sound/dsp crw-rw 1 almaw audio 14,3 Jan 1 1970 /dev/sound/dsp But I suspect that your real problem is that in addition to the lines you specified in modules.d/alsa, you must have the following: alias snd-card-0 snd-via82xx -- replace with your ALSA driver alias snd-slot-0 snd-card-0-- required for OSS support under ALSA Regards, Alastair ___ 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] Asterix Sip Stack
Mark Phillips wrote: [EMAIL PROTECTED] said: Is it Sip Registry Server ?. Could it work as Proxy Server ? Hello Ahmet, Asterisk is more than a proxy. Its an entire PBX. At a basic level it can be used as a proxy though. My favourite subject... :-) No, Asterisk is not even close to a SIP proxy. It's a PBX that supports SIP. It is also a SIP registrar. If you go through the archives, you have much longer explanations there on this particular topic. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Schools/Districts using asterisk?
On Thu, 18 Mar 2004, Chris Hobbs wrote: I'm investigating asterisk to use as a replacement for an aging Lucent PBX in our district office, as well as replacing the Centrex/intercom based systems at our schools. I'm curious if any other schools/districts are using asterisk? If so, I'd certainly be interested in talking about your setup. Chris, We're in the process (final phase to be completed in the next two weeks) of replacing our ATT/Lucent/Avaya solution at our high school. Over the summer, we hope to integrate/replace the PA/intercom system (Teltrend IV) to extend asterisk dialtone to the classrooms. Contact me off list and I'd be more than happy to talk with you about our progress, plans, and experiences. 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
[Asterisk-Users] T100P and outbound calls.
Hey all. We've just recently purchased a T100P in order to provide VoIP to a remote office. We've interfaced it with a DS1-formatter on our Mitel GX5000 switch. I realize that plugging the * PBX into this class 5 switch isn't the best situation to have in the world...but hey it's what we've got. We've just been having a couple of problems which I believe to be related, but can't figure out. First of all, outgoing calls fail. The * box shows that it is dialing out, assigned to the proper group, however I never hear anything. The call will sit there idle as long as I allow it to. If I look at the status on the GX5000, it shows that channel as being blocked, but will go back to idle the moment I hang up. Unfortunately I can't (it's possible but requires equipment that we don't have) monitor the GX to see what is happening when I place the call from the * box. The second problem, which I think to be caused by the same issue, has to do with hearing a ring. When placing a call from the outside that travels over that T1, to the * box, I can see the call being directed to the proper SIP location, it shows the phone as ringing, and indeed the phone at the receiving end of the line is ringing. However the caller does not hear the ring...only dead air. While I know this is not a major issue, I'm sure it will lead some to believe that the call is not getting through since they don't hear the ringing on their phone. No I don't have a ton of experience with this stuff...but I am picking up a lot of concepts rather quickly...so please bear with me. Thanks for your help. Mark ___ 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: Random Echo
TC, Thanks for your recommendation. Looking at sourcing one now. This is great news. As I understand it, you need the card, Chasis, and Power Module, and we should be up and running? Thanks, Brent -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kekin Dand Sent: Thursday, March 18, 2004 1:06 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Re: Random Echo TC, Appreciated your help and will try out TelLabs card and see if we can get rid of echo. Yesterday I did some changes in TX and RX attenuation setting on Channel bank and it reduces the echo, but it is not yet vanished as we wanted. Any way Thanks. Regards, KD Date: Wed, 17 Mar 2004 19:47:16 -0800 From: TC [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Random Echo To: [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] I did some google search but didn't find any details, about how to configure between Adtran 750 and T100P. If you have already done, please give us some details. not sure what level of dtl you want its quite straight fwd It varies depending on the chasis but in general there are T1 in and T1 out DB-15's for each T1 circuit you want to echo cancel. A straight T1 rj-45 cable goes to the channel bank other end is db-15 in to Tellab then a T1 X over goes from the Tellab db-15 out to the T100p card Then there are external Mode and Chan switchs that allow you to configure the T1 circuit (the line bld, framing, and coding/signaling), and then other setting to allow channels FXO/FXS LS, GS and enable echo cancel channel by channel, I do it for all fxs/fxo ports and turn ALL * echo cancel in zapata OFF.. ___ 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] Softfax/spandsp
Dear Steve sorry for my last bug report (not full reported well) I have now reinstall your spandsp 1b but i have this type of error... - Test Fax station Canon B150 - -- Executing RxFAX(Zap/33-1, /home/user/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 37 30 38 30 35 33 31 33 33 30 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 0331350807 DCS: 83 00 86 a0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 10ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1684.65 (18) Fast carrier down Fast carrier up Coarse carrier frequency 1699.39 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.10 (4917) Fast carrier down Fast carrier up Coarse carrier frequency 1699.55 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.10 (4916) Fast carrier down Fast carrier up Coarse carrier frequency 1699.27 (86) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.09 (2925) Fast carrier down Fast carrier up Fast carrier down -- Executing Hangup(Zap/33-1, ) in new stack --- Thanks for great work!! Dimitri On Thursday 18 March 2004 14:06, Steve Underwood wrote: Hi all, It seems this week's release of spandsp fixed the major problems in the previous release, but still people have had a lot of trouble. Working with some of those who tried the software and gave me good feedback, I have identified some apparently common bugs in fax machines, and I have implemented workarounds for these in spandsp, and feedback so far seems good. I also fixed a couple of bugs. I think this version will work proper with a much wider range of fax machines. However, people have warning me that fax machines have a bad habit of not following the specs properly :-( There is now a new tarball at ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.1b.tar.gz Please try this, and report any problems you find. This version has the following changes: A floating point exception has been fixed A problem with the software not properly Some fax machines send a little less than the specified 1.5 seconds of training test data, so the training test failed every time. I now only look for 1.25seconds of training test data. Some fax machines do not correctly initialise the scrambler in their V.29 transmit modem. I have changed the software to tolerate this. Some fax machines send a burst of ones before the burst of zeros that forms the training test data. I have changed the software to tolerate this. 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 ___ 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] Schools/Districts using asterisk?
Something you should look at in a school setting is the Wisip wireless 802.11b phone from Pulver. We are looking at this to help reduce cellular costs and also as a possible teacher phone. Not unless transfers aren't important to you. I have been unable to determine how to initiate a transfer with this phone. The selection of ringtones blows donkeys too. Can anyone explain why the hell it seems impossible to find a VOIP phone with a good selection of NORMAL (5 years ago) cell-style ringtones? Not tunes, not a POTS ringback tone, but a good selection of shrill, easy to identify and businesslike ringing tones???!?!?!?? Regards, Andrew ___ 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] Monastery Devel snapshot
I'll answer my own question ... If you don't call the database asterisl you need to edit in the name you do use to status.php otherwise monastery behaves as though nothing is happening rather than flagging an error ;-) Iain --On Thursday, March 18, 2004 5:51 pm + Iain Stevenson [EMAIL PROTECTED] wrote: ... just installed this. The database updates OK but status.php shows no active channels (either SIP to SIP or SIP to voicemail). Iain --On Thursday, March 18, 2004 11:51 am -0500 Tim Sailer [EMAIL PROTECTED] wrote: I just pushed out a snapshot of the -devel version of monastery. ftp://ftp.buoy.com/pub/asterisk/monastery-devel.tgz Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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] Explain ring tones (was Schools/Districts using asterisk?)
On Thu, 2004-03-18 at 15:38, Andrew Kohlsmith wrote: snip Can anyone explain why the hell it seems impossible to find a VOIP phone with a good selection of NORMAL (5 years ago) cell-style ringtones? Not tunes, not a POTS ringback tone, but a good selection of shrill, easy to identify and businesslike ringing tones???!?!?!?? Regards, Andrew Jeff Pulver himself said at the Autumn VON Don't trust a product manager over 30! This is to say if what you want isn't trendy and hot, it ain't happnin' Pass the bromo-seltzer... Howard White ___ 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] Problems with asterisk and gnophone on Gentoo box
On Thursday 18 March 2004 14:57, Alastair Maw wrote: On 18/03/04 18:46, Kevin wrote: Thanks for your reply, Alastair. I did use that guide in getting myself set-up with sound, and do have alsa-oss installed: You need to have it all insmod'ed as well (which I guess it will be): [EMAIL PROTECTED] almaw # lsmod | grep oss snd-seq-oss29216 0 snd-seq-midi-event 3584 0 [snd-seq-oss] snd-seq37584 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4304 0 [snd-rawmidi snd-seq-oss snd-seq] snd-pcm-oss38436 0 snd-pcm60960 0 [snd-via82xx snd-pcm-oss] snd-mixer-oss 13680 0 [snd-pcm-oss] snd33636 1 [...snip...] Yep. I have this, or something very close to it anyway: bash-2.05b# lsmod | grep oss snd-pcm-oss39140 0 (unused) snd-pcm65828 0 [snd-pcm-oss] snd-mixer-oss 13392 0 [snd-pcm-oss] snd-seq-oss27456 0 (unused) snd-seq-midi-event 3840 0 [snd-seq-oss] snd-seq40528 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4176 0 [snd-seq-oss snd-seq] snd33892 0 [snd-pcm-oss snd-pcm snd-mixer-oss snd-seq-oss snd-seq-midi-event snd-seq snd-timer snd-seq-device] I wonder if the (unused) messages are telling me something important here... I see that your output does not have them, apparently indicating that something is using them. In addition to the sound apps I have that use alsa, I also use xmms with a libOSS.so plugin for accessing the oss system. xmms does work for me with this plugin (and doesn't when I use the libALSA.so plugin). Is it safe to conclude therefore, that xmms _is_ properly accessing the OSS emulation support in the alsa system with this libOSS.so plugin? If so, is it safe to conclude that my OSS emulation is working properly? Also make sure your dsp device is accessible for the user running OSS: [EMAIL PROTECTED] almaw # ls -l /dev/dsp lr-xr-xr-x 1 root root 9 Mar 9 10:02 /dev/dsp - sound/dsp [EMAIL PROTECTED] almaw # ls -l /dev/sound/dsp crw-rw 1 almaw audio 14,3 Jan 1 1970 /dev/sound/dsp Yup, have that too. And just to make sure, I'm running asterisk and gnophone as root. I've had success running asterisk v1-0_stable from CVS as root on other linux distributions like SuSE 9.0, but I think it doesn't use alsa---I think it uses straight OSS---not sure though. But I suspect that your real problem is that in addition to the lines you specified in modules.d/alsa, you must have the following: alias snd-card-0 snd-via82xx -- replace with your ALSA driver alias snd-slot-0 snd-card-0-- required for OSS support under Ah! Though I missed it with my grep in my original reply, I followed up with another that showed them being present (about 10 minutes before you posted---probably not on the list yet). But your post here shows me that I had a syntax problem in my config file. Whereas I had: alias sound-slot-0 snd-card-0 ^^ I obviously should have had: alias snd-slot-0 snd-card-0 That certainly helps (or I think it should anyway). Unfortunately, after fixing this flaw in the config file and rebooting in order to reload all the modules, I still have the same problems (with both gnophone and asterisk). I ran modules-update, checked /etc/modules.conf for proper carryover of these changed settings and rebooted again, but still no joy. Asterisk startup output: * [app_waitforring.so] = (Waits until first ring after time) == Registered application 'WaitForRing' [app_setcidnum.so] = (Set CallerID Number) == Registered application 'SetCIDNum' [chan_oss.so] = (OSS Console Channel Driver) Mar 18 16:39:40 WARNING[16384]: chan_oss.c:352 setformat: Requested 8000 Hz, got 7866 Hz -- sound may be choppy Mar 18 16:39:40 WARNING[16384]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Mar 18 16:39:40 WARNING[229391]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable [app_db.so] = (Database access functions for Asterisk extension logic) == Registered application 'DBget' == Registered application 'DBput' == Registered application 'DBdel' == Registered application 'DBdeltree' [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI *CLI * Where's that comment about
[Asterisk-Users] zaphfc problem
Hi,
I have a partial working installation with zaphfc.
Incoming call :
For incoming call, seems work fine. But the sound is very bad with bounce
short crashing sound. Same sound with echo cancel off or on.
SDA work fine.
Another problem, it's seems that's zaphfc don't reset correctly the line. I
have one of my D channel how was busy even after stop communication.
Outgoing call :
When try make a call, i have error like this :
Mar 18 22:44:05 WARNING[229391]: chan_zap.c:5952 zt_pri_error: PRI: !! Got
reject for frame 1, but we have nothing -- resetting!
MFE for TEI = 80
== D-Channel on span 1 up
== D-Channel on span 1 down
== D-Channel on span 1 down
Config is mostly like howto on voip-info.org
in /var/log/messages, i have hundred of this line :
zaphfc: empty HDLC frame received
---
Hardware : Bewan Gazel PCI (have his dedicaced IRQ)
---
ztcfg :
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
Channel map:
Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
---
/etc/zaptel.conf :
span=1,1,3,ccs,ami
bchan=1-2
dchan=3
fxsks=4
---
/etc/asterisk/zapata.conf :
[snip]
switchtype = euroisdn
signalling = bri_cpe_ptmp
pridialplan=local
echocancel=yes
immediate=yes
;setcallerid(${CALLERIDNUM})
;usecallerid=yes
group = 1
context=incoming
channel = 1-2
[snip]
Don't work with bri_net_ptmp
---
ISDN operator : France Telecom
---
*CLI zap show channel 1
Channel: 1
File Descriptor: 25
Span: 1
Extension: s
Context: incoming
Caller ID string: xx
Destroy: 0
Signalling Type: PRI Signalling
Owner: Zap/1-1
Real: Zap/1-1
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
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
PRI Flags: Call
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
When offline :
[snip]
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
PRI Flags:
Mar 18 22:40:57 WARNING[16384]: chan_zap.c:7055 zap_show_channel: Failed to
get conference info on channel 1
Mar 18 22:40:57 WARNING[16384]: chan_zap.c:7061 zap_show_channel: Failed to
get confmute info on channel 1
Thanks for help !
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
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[Asterisk-Users] CCM - GnuGK - *
I've got my GnuGK box listening on 1720.. CCM thinks it's an OH323 gateway with 245 tunneling... * is registering the # as being an extension in GnuGK.. I call the # and I see the port 1720 light up with tcpdump.. but gk -ttt isn't showing my anything and nothing gets send to the * box... What am I missing? Kyle ___ 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] Speaking of ring tones...
Anyone know if Grandstream ever plan to implement another tone on the BT-101? To me, it's very weird hearing ringback as the ring-in sound. Cheers, Chris. ___ 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] Explain ring tones
Can anyone explain why the hell it seems impossible to find a VOIP phone with a good selection of NORMAL (5 years ago) cell-style ringtones? Not tunes, not a POTS ringback tone, but a good selection of shrill, easy to identify and businesslike ringing tones???!?!?!?? No kidding. Imagine my surprise when it sounded like my VOIP phone was dialing out on its own (The Grandstream phones use a POTS ringback tone). I picked up the handset and there was someone on the line! How did that even make sense to Grandstream? Why would you want your phone to use the same sound for a call coming in as a call going out?? -- Kevin Williams Senior Developer Applianz Technologies, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * and PrePaid
There is no mention of * there at all or maybe I am blind :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Frank Norman Sent: Friday, 19 March 2004 1:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * and PrePaid There is a configuration and billing system for *. Refer to www.vidanetwork.com Do you Yahoo!? Yahoo! Mail - More reliable, more storage, less spam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk AGI and DTMF
All, I have my AGI working. I am placing a call in the outgoing directory and running my AGI. Once the call is places and answered I then need to send DTMF tones. Like 101. How can I do this in the AGI? I did not see and commands for it. Thanks, Jerry ___ 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] Softfax/spandsp
Title: RE: [Asterisk-Users] Softfax/spandsp Hi, We've tried J2 faxing with the newest release of spandsp and found the same issue as with our own Dialogic based faxing. Steve, we can fax to your system from our platform and/or J2 if you think it can help. Here is what we see on incoming fax from J2: -- Executing RxFAX(Zap/28-1, /usr/tmp/nativefax-from-2132255675-1079649839.tif) in new stack Changed from phase 0 to 1 Slow carrier up [03/18/04 16:43:59.245] DEBUG[655386]: File chan_zap.c, Line 3332 (zt_exception): Exception on 43, chann el 28 [03/18/04 16:43:59.245] DEBUG[655386]: File chan_zap.c, Line 2753 (zt_handle_event): Got event No event( 0) on channel 28 (index 0) [03/18/04 16:43:59.246] DEBUG[655386]: File chan_zap.c, Line 3234 (zt_handle_event): Dunno what to do wi th event 0 on channel 28 Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 39 30 34 38 39 31 39 37 34 38 31 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 18479198409 DCS: 83 00 06 f0 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1709.56 (90) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1702.22 (86) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1699.93 (5195) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1702.50 (85) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1699.92 (5195) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1702.55 (85) Fast carrier training failed Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1700.12 (3083) Fast carrier down 0 bad bits in trainability test FTT: 44 ^M -- Channel 4, span 2 got hangup [03/18/04 16:44:28.468] DEBUG[655386]: File app_rxfax.c, Line 200 (rxfax_exec): Got hangup Thank you. Alex Zarubin Webley Systems -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED]] Sent: Thursday, March 18, 2004 8:07 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Softfax/spandsp Hi all, It seems this week's release of spandsp fixed the major problems in the previous release, but still people have had a lot of trouble...
[Asterisk-Users] PRI Errors
Can anyone decipher these error messages? Mar 18 18:10:21 WARNING[131081]: chan_zap.c:5949 zt_pri_error: PRI: Read on 39 failed: Unknown error 500 Mar 18 18:10:21 NOTICE[131081]: chan_zap.c:6664 pri_dchannel: PRI got event: 6 on span 1 Thanks, TL ___ 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] Softfax/spandsp - page cut-off
Title: RE: [Asterisk-Users] Softfax/spandsp - page cut-off Hi, Faxing from Dialogic and J2 is one problem. Another problem is a page cut-off - happened 2 times out of 7-8 when faxing from the real fax machine. ... Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1703.83 (149) Fast carrier down 0 bad bits in trainability test FTT: 44 Fast carrier up Coarse carrier frequency 1700.18 (86) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.35 (89) Fast carrier trained Fast carrier down Fax3Decode2D: (FakeInput): Bad code word at scanline 2744 (x 1456). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2744 (got 1456, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2745 (x 1006). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2745 (got 1006, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2746 (got 1796, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2747 (got 2866, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2748 (got 1732, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2749 (got 2091, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2751 (got 2578, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2752 (x 834). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2752 (got 834, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2753 (got 2389, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2754 (got 2863, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2755 (got 2180, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2756 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2757 (got 2033, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2758 (x 313). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2758 (got 313, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2759 (got 1790, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2760 (x 4). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2760 (got 4, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 2761 (x 898). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2761 (got 898, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 2763 (x 1626). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2763 (got 1626, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2766 (got 2536, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2767 (x 155). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2767 (got 155, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2769 (got 2257, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2770 (got 1756, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 2771 (x 368). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2771 (got 368, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2772 (got 2129, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2773 (got 2535, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2774 (got 2063, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2775 (got 1790, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 2776 (x 88). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2776 (got 88, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2777 (got 2433, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2779 (got 3158, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2780 (got 2186, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2781 (got 2275, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 2782 (got 646, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2783 (got 2240, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 2784 (got 2273, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length
[Asterisk-Users] Problems with FWD
Hi Folks, Anyone having issues with FWD lateley? It seems that ever since they sent me a notification about my voicemail I've been unable to sucessfully make calls to my WA phone number which is forwarded to FWD. Also, on my office machine I'm unable to properly register with FWD. I get a lot of back and forth traffic which terminates with this; Sip read: SIP/2.0 200 Recieved private address, use public IP next time Via: SIP/2.0/UDP 192.168.18.65:5060;branch=z9hG4bK048441f0;received=192.168.18.65 From: sip:[EMAIL PROTECTED];tag=as2e4632d5 To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.777f Call-ID: [EMAIL PROTECTED] CSeq: 117 REGISTER Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 I have the following in my sip.conf file; register = 248249:[EMAIL PROTECTED]/3409 ; FreeWorldDialup account [fwd] type=friend secret=blueroyal username=248249 host=fwd.pulver.com nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw mailbox=3409 My machine is behind a Checkpoint firewall. Its public address 63.88.139.198; private address is 192.168.18.65. All the normal ports are open. 5000-6000 1-2. Ideas? Mark -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Problems with asterisk and gnophone on Gentoo box
What sound chip are you using? I thought I had the via82xx and spent a couple days jacking with it before I figured out I was wrong. Here's my alsa setup in modules.conf: # --- ALSACONF verion 1.0.0pre1 --- alias char-major-116 snd alias char-major-14 soundcore alias char-major-15 off alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss alias snd-card-0 snd-intel8x0 alias sound-slot-0 snd-intel8x0 # --- END: Generated by ALSACONF, do not edit. -- I'm thinking maybe soundcore is what you're missing, since on mine it's definitely used. As proof, here's the pertinent readoff from lsmod: snd-mixer-oss 13456 0 (autoclean) [snd-pcm-oss] snd-intel8x0 20612 1 snd-ac97-codec 50176 0 [snd-intel8x0] snd-pcm78464 0 [snd-pcm-oss snd-intel8x0] snd-page-alloc 8876 0 [snd-intel8x0 snd-pcm] snd-timer 19204 0 [snd-pcm] snd-mpu401-uart 4856 0 [snd-intel8x0] snd-rawmidi17728 0 [snd-mpu401-uart] snd-seq-device 5644 0 [snd-rawmidi] snd42468 0 [snd-pcm-oss snd-mixer-oss snd-intel8x0 snd-ac97-codec snd-pcm snd-timer snd-mpu401-uart snd-rawmidi snd-seq-device] soundcore 6244 4 [snd] What motherboard are you using? Again, make sure you've got the right chip selected for alsa. John On Thu, 2004-03-18 at 16:11, Kevin wrote: On Thursday 18 March 2004 14:57, Alastair Maw wrote: On 18/03/04 18:46, Kevin wrote: Thanks for your reply, Alastair. I did use that guide in getting myself set-up with sound, and do have alsa-oss installed: You need to have it all insmod'ed as well (which I guess it will be): [EMAIL PROTECTED] almaw # lsmod | grep oss snd-seq-oss29216 0 snd-seq-midi-event 3584 0 [snd-seq-oss] snd-seq37584 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4304 0 [snd-rawmidi snd-seq-oss snd-seq] snd-pcm-oss38436 0 snd-pcm60960 0 [snd-via82xx snd-pcm-oss] snd-mixer-oss 13680 0 [snd-pcm-oss] snd33636 1 [...snip...] Yep. I have this, or something very close to it anyway: bash-2.05b# lsmod | grep oss snd-pcm-oss39140 0 (unused) snd-pcm65828 0 [snd-pcm-oss] snd-mixer-oss 13392 0 [snd-pcm-oss] snd-seq-oss27456 0 (unused) snd-seq-midi-event 3840 0 [snd-seq-oss] snd-seq40528 2 [snd-seq-oss snd-seq-midi-event] snd-seq-device 4176 0 [snd-seq-oss snd-seq] snd33892 0 [snd-pcm-oss snd-pcm snd-mixer-oss snd-seq-oss snd-seq-midi-event snd-seq snd-timer snd-seq-device] I wonder if the (unused) messages are telling me something important here... I see that your output does not have them, apparently indicating that something is using them. In addition to the sound apps I have that use alsa, I also use xmms with a libOSS.so plugin for accessing the oss system. xmms does work for me with this plugin (and doesn't when I use the libALSA.so plugin). Is it safe to conclude therefore, that xmms _is_ properly accessing the OSS emulation support in the alsa system with this libOSS.so plugin? If so, is it safe to conclude that my OSS emulation is working properly? Also make sure your dsp device is accessible for the user running OSS: [EMAIL PROTECTED] almaw # ls -l /dev/dsp lr-xr-xr-x 1 root root 9 Mar 9 10:02 /dev/dsp - sound/dsp [EMAIL PROTECTED] almaw # ls -l /dev/sound/dsp crw-rw 1 almaw audio 14,3 Jan 1 1970 /dev/sound/dsp Yup, have that too. And just to make sure, I'm running asterisk and gnophone as root. I've had success running asterisk v1-0_stable from CVS as root on other linux distributions like SuSE 9.0, but I think it doesn't use alsa---I think it uses straight OSS---not sure though. But I suspect that your real problem is that in addition to the lines you specified in modules.d/alsa, you must have the following: alias snd-card-0 snd-via82xx -- replace with your ALSA driver alias snd-slot-0 snd-card-0-- required for OSS support under Ah! Though I missed it with my grep in my original reply, I followed up with another that showed them being present (about 10 minutes before you posted---probably not on the list yet). But your post here shows me that I had a syntax problem in my config file. Whereas I had: alias sound-slot-0 snd-card-0 ^^ I obviously should have had: alias snd-slot-0 snd-card-0 That certainly helps (or I think it should anyway). Unfortunately, after fixing this flaw in the config file and rebooting in order to reload all the modules, I still
Re: [Asterisk-Users] help me: warnings on Read error on sound device, Ignoring rxwink
Michael, As far as I'm aware, RedHat 9 uses the ALSA sound drivers. you need to prevent asterisk from loading the OSS channel driver with: noload = chan_oss.so in your modules.conf -Ben -- Computer games do not affect kids! If Pac-Man had effected us as kids then we would now be running around in darkened rooms dancing to repetitive music and munching pills. ___ 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: Text message
___ 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] Problems with FWD
From what I know is that if you have a private IP on your asterisk box, and only a private IP, your box will send out SIP messages containing your private IP in the FROM field. try to add this in your sip.conf externip=63.88.139.198 David - Original Message - From: Mark Phillips [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 18, 2004 3:16 PM Subject: [Asterisk-Users] Problems with FWD Hi Folks, Anyone having issues with FWD lateley? It seems that ever since they sent me a notification about my voicemail I've been unable to sucessfully make calls to my WA phone number which is forwarded to FWD. Also, on my office machine I'm unable to properly register with FWD. I get a lot of back and forth traffic which terminates with this; Sip read: SIP/2.0 200 Recieved private address, use public IP next time Via: SIP/2.0/UDP 192.168.18.65:5060;branch=z9hG4bK048441f0;received=192.168.18.65 From: sip:[EMAIL PROTECTED];tag=as2e4632d5 To: sip:[EMAIL PROTECTED];tag=b27e1a1d33761e85846fc98f5f3a7e58.777f Call-ID: [EMAIL PROTECTED] CSeq: 117 REGISTER Server: Free World Dialup (0.8.11rc3 (i386/linux)) Content-Length: 0 I have the following in my sip.conf file; register = 248249:[EMAIL PROTECTED]/3409 ; FreeWorldDialup account [fwd] type=friend secret=blueroyal username=248249 host=fwd.pulver.com nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw mailbox=3409 My machine is behind a Checkpoint firewall. Its public address 63.88.139.198; private address is 192.168.18.65. All the normal ports are open. 5000-6000 1-2. Ideas? Mark -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 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] * and IConnectHere
Hi to everyone When I dial a phone numer using my IConnectHere acount I get this message. Can someone tell me what it is? Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:37 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424 ast_dsp_process: Unable to process inband DTMF on 1 frames Mar 18 18:59:38 WARNING[1217602880]: dsp.c:1424
[Asterisk-Users] Cisco 7960 SIP Firmware
Out of everyone using the 7960 currently, what would you say is the best firmware to use w/ asterisk? What's the most compatible / stable? In addition, is there a better / easier or straightforward tutorial to upgrading the firmware? 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
[Asterisk-Users] loopstart,kewlstart,groundstart
kindly tell me what is difference b/w loopstart, kewlstart, groundstart for FXO or FXS devices Thank you -- Atif Rasheed Convergence (Business Systems) http://www.convergence.com.pk -- ___ 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] Project
I want to set up a Linux answering machine/voice mail deal that will e-mail me phone messages. I looked into the voice modem (vgetty) stuff, but I'm not getting a warm fuzzy feeling about it from reading the mailing lists - much of it old and not encouraging - and it seems that there is not a lot of manufacturers of voice modems today etc. I currently have a Panasonic KX-T616 phone system and the hand sets. Are there any Asterisk systems that would support them? I could just upgrade to a newer digital Panasonic system and continue to use the handsets (there is a serial interface I could talk to via linux)? Perhaps all I need is some type of phone card that can act as an answering machine connection? I am also looking into the possibility of replacing the Panasonic equipment with an asterisk system - but I have some doubts. It seems to me that a PBX should have a RTOS that Linux might talk to - my background is EE with work with embedded micros so I realize there are limitations in using linux as the switch. My needs are humble - I have 3 CO lines and 16+ extensions - I would like to end up with something OSS/GPL if I can. It would be nice to detect Fax calls and run caller ID to the LCD on the telephones. Should I be looking at Asterisk? What hardware should I consider? I wouldn't be against paying someone on this list for a 1/2 hour phone consultation if you know this stuff inside out. I also humbly worry about maintaining something written in perl, as perl is a write only programming language in my experience. Most any other language would be easier to maintain (I've written a few things in perl - there is nothing quite like coming back and looking at my own perl code from a few years ago and finding I don't a clue as to what I was doing with less than a exhaustive inspection.) -- -- Karl Schmidt EMail[EMAIL PROTECTED] Transtronics, Inc.WEB http://xtronics.com 3209 West 9th Street Ph(785) 841-3089 Lawrence, KS 66049FAX(785) 841-0434 When angry count four; when very angry, swear. --Mark Twain - ___ 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] Speaking of ring tones...
I kinda like it .. ;) Nice conservative. OTOH, the new snom 200 I just got today has some reeeaaally weird ring tones (and nothing really 'traditional'). Now, maybe we should take a lesson from the cell-phone people, and talk manufacturers into letting us download ringtone(s). Cheers, WW - Original Message Follows - Anyone know if Grandstream ever plan to implement another tone on the BT-101? To me, it's very weird hearing ringback as the ring-in sound. Cheers, Chris. ___ 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 Willy Wouters ypOne Publishing ___ 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] Cisco IOS crash with multiple SIP endpoints behind NAT
Thought I'd drop a note here in case anyone else has been experiencing this. Cisco routers running NAT are liable to crash when doing NAT translation of SIP packets. Symptoms are a watchdog caused software abort in the IP Input process. The router then reloads. In my case this has been happening every ~24 hours with one SIP device behind the NAT and every ~15 minutes with three devices. Cisco have identified this as bug ID CSCed42990 (http://www.cisco.com/cgi-bin/Support/Bugtool/onebug.pl?bugid=CSCed42990) but it is not yet fixed in a released version. You can get an interim release from TAC if you have a service contract. According to the bug description this only affects 12.3, but it definitely happened in the 12.2T train too (this is why we upgraded to 12.3). David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Schools/Districts using asterisk?
Chris, I read your message below, posted to the Asterisk-Users board. I don't represent or currently have a customer with an asterisk installation in a school district environment, but if you have any questions, I'd be happy to chat with you. I'm a network and asterisk private consultant. I've been developing asterisk based phone installations for 2 years. However any chat time is free as the information you provide me as to your needs and requirements outweighs any information I might share with you during the call. In the end if I am of any help and you look to get outside help for an installation should you choose to do one, I'd appreciate letting me bid on the project, but besides that, no strings attached to our conversations Thank you, On 11:21 AM 3/18/2004, Chris Hobbs wrote: I'm investigating asterisk to use as a replacement for an aging Lucent PBX in our district office, as well as replacing the Centrex/intercom based systems at our schools. I'm curious if any other schools/districts are using asterisk? If so, I'd certainly be interested in talking about your setup. Thanks, -- Chris Hobbs Silver Valley Unified School District Head geek: Technology Services Coordinator webmaster: http://www.silvervalley.k12.ca.us/~chobbs/ postmaster: [EMAIL PROTECTED] pgp: http://www.silvervalley.k12.ca.us/~chobbs/key.asc Chris A. Icide 332 Valdez Ave. Half Moon Bay, CA 94019 650-712-8223 voice 650-712-8995 fax ___ 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] Schools/Districts using asterisk?
Ack, sorry for the reply to the message board on my last reply to this topic, I forgot to replace the To: person. Thanks for the replies in advance telling me that that wasn't on topic, I consider myself in error, and offer the appropriate apology. -Chris ___ 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] Redhat lastest kernel problems with Zaptel drivers kernel 2.4.20-30.9
I ran the up2date and installed newest kernel 2.4.20-30.9 and rebooted the modules for the zaptel drivers wcfxo and wcfxs didn't load I reboot with kernel 2.4.20-28.9 and all is working fine. I didn't have time to work in the issue but have seen it on two systems. Anyone have any idea what the issue is? 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] Cisco 5350 One Way Sound
Hello All! I have successfully set up my Cisco 5350 for use with *! Through direct-inward-dial i have all my users dialing my number placed in Asterisk. But I have a problem - one way sound (it IS NOT a codec issue): When I call the 5350, it connects to the Asterisk, and then to the destination. I can hear the other party, but they can't hear me. I tried all codecs possible (G.711a/u, G729). I've done the tests with ATA-186 connectedto my Asterisk, using the same codec as the 5350. Any ideas? Thank you for your time in advance! Greetings, Doichin Dokov NetOne - Silistra ___ 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
