[Asterisk-Users] Newbie ... some questions
Hi guruz, I haverequirements from a company, which isgoing to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase thehardware they want if following is possible in the asterisk software 1) they want to whisper withone side of the call i.e. the manager while monitoring thecalls (from customers to support staff) from his extension (either SIP or Zap Channel),canguide support personif he is in trouble talking with the client 2) while monitring the call, incharge can take the call and start talking with the customer directly and the support officer gets a hangup tone Plz suggest me if the above is possible and how the above can be achieved. TIA Franzi Post your free ad now! Yahoo! Canada Personals
Re: [Asterisk-Users] Newbie ... some questions
Franz S wrote: Hi guruz, I have requirements from a company, which is going to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase the hardware they want if following is possible in the asterisk software 1) they want to whisper with one side of the call i.e. the manager while monitoring the calls (from customers to support staff) from his extension (either SIP or Zap Channel), can guide support person if he is in trouble talking with the client 2) while monitring the call, incharge can take the call and start talking with the customer directly and the support officer gets a hangup tone Plz suggest me if the above is possible and how the above can be achieved. TIA Franzi I seriously doubt these things are possible.. not without recoding some of the Asterisk components.. Later ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie ... some questions
Hmmm any comments from Digium ... ? FranziWipeOut [EMAIL PROTECTED] wrote: Franz S wrote: Hi guruz, I have requirements from a company, which is going to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase the hardware they want if following is possible in the asterisk software 1) they want to whisper with one side of the call i.e. the manager while monitoring the calls (from customers to support staff) from his extension (either SIP or Zap Channel), can guide support person if he is in trouble talking with the client 2) while monitring the call, incharge can take the call and start talking with the customer directly and the support officer gets a hangup tone Plz suggest me if the above is possible and how the above can be achieved. TIA FranziI seriously doubt these things are possible.. not without recoding some of the Asterisk components..Later___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Free Pop-Up Blocker - Get it now
Re: [Asterisk-Users] The internet needs a dialing code..
Even more cool is to start using ENUM. There's a good new article on how to start doing that on the Wiki, not contributed by me. Since the ENUM tree is not very active, only experiments in some countries, we could start building our own Asterisk/IAXtel ENUM-like tree. One problem though is that if you call me by ENUM, you'll get a SIP URL that your Asterisk won't be able to handle. As soon as outbound SIP url is fixed, as well as calling IAX by URL, we'll be able to use Asterisk to * Automatically off-load outbound calls to VoIP (ENUM will tell if the phone number you're trying to call is available on the net through SIP, H.323 or IAX) * Connect our IAX servers and networks automagically. ENUM is a magic solution for building structures like the one you're suggesting, WipeOut. The article http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing Reference on the EnumLookup command with examples from nic.at enum trials: http://www.voip-info.org/wiki-Asterisk+cmd+EnumLookup /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
The amount of mail on asterisk-users is more than even *I* can read in a day, and my job is 100% asterisk. There probably is a justification for a new list, but I think it is less the -biz list as much as much as the -newbies. Keeping a business discussion on -users is probably quite useful since often times a business discussion can involve technical details of what Asterisk is capable of doing. I propose Asterisk-intro where people new to Asterisk can ask questions in an open manner and people with experience jump in when they have time to answer, guide and help. A list with open attitude and guidance. There's enough of us that from time to time have time :-) to assist, but at some times want to concentrate on a higher-level discussion, helping each other on a professional level with Asterisk. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The internet needs a dialing code..
Internationally, there is already an officially sanctioned country code for Universal Telecommunications Services, and it's +878. There is quite a bit of activity now in moving that area code from the ITU sanctioning (which happened a few weeks ago) and now moving towards commercial implementation. http://www.visionng.org/index.htm pulver.com is involved in this organisation. In addition to this work, some countries have added non-geographic area codes. In Sweden 075 is designated as a personal non-geographic area code. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
[201]
Username=davy
Technology=SIP
DeviceID=davy
[202]
Username=pieter
Technology=SIP
DeviceID=pieter
201 is the extension from in extensions.conf
davy = the thing between brackets in sip.conf
When i try to click on one of the red boxes in the manager, i always get:
Event: Status -|- Channel: SIP/davy-1a07 -|- CallerID: davy -|- State: Up
-|- Context: sip -|- Extension: 202 -|- Priority: 1 -|- Link:
SIP/pieter-e582 -|- Uniqueid: 1069581.102
Response: Error -|- Message: Invalid channel
At 21:06 21/11/2003 -0600, you wrote:
Here's the structure for the monitor.conf file:
[1101] Extension Number (from extensions.conf in
Asterisk)
UserName=Blah Blah Label. Simply sets the caption for the button.
Technology=SIP Technology used for stations (SIP, MGCP, Zap,
etc.)
DeviceID=1101 Device identifier (from sip.conf in this case)
All of the Technology values are normal asterisk values except for APP,
which is an application (like Voicemail or MOH or MeetMe) and PSTN,
which is a number outside of the Asterisk inside dial plan.
I hope this helps. Remember that for PSTN and APP values, the bracketed
Extension number and the DeviceID need to be the same.
Regards,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of zoa
Sent: Friday, November 21, 2003 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
(Alpha)
Could you give me some explanation on how to use the configuration
file ?
I always get INVALID channels if i click on the red icon next to my
name.
(Maybe i should use a numeric context or use numeric user names?)
Tool looks great, this will be a very cool asterisk addition.
zoa.
At 16:43 21/11/2003 -0600, you wrote:
I think the script host gets installed with Windows explorer. If you
don't have it, you can use the DLL in the dlls download:
http://www.sokol-associates.com/Downloads/Dlls.zip
Hope that helps.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walker Haddock
Sent: Friday, November 21, 2003 4:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Call Manager for Windows
0.0.1
(Alpha)
here:
http://www.sokol-associates.com/Downloads/AstMgr.zip
It's written in VB6 (yes - barf, gag, whatever). The only thing
required beyond the integral VB6 controls is the Windows
Scripting
Runtime which most PCs should have. I will work on an
installable
version soon. I may also port it to something more
cross-platform.
Please bear with me as I am just learning Gnome/GTK/X-windows.
Steve, how do you know if the Windows Scripting Runtime is
installed
in
Windows XP Pro?
Where do you get it from and how should it be installed?
Thanks, Walker
--
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**
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DataCrest, Inc.e-mail: [EMAIL PROTECTED]
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Re: Asterisk Lists (was Re: [Asterisk-Users] Asterisk Business discussion again)
On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote So far it seems like the proposed candidates for new lists are: asterisk-newbies (perhaps a better word?) Maybe asterisk-install ? asterisk-starters ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
I will say that the Wiki is very hard to deal with as getting information out of it! It tends to go in the wrong direction allot! Some of us just don't have the time to go through it! Could you please elaborate a bit more, to help us steer the wiki in the right direction? As I see it, the Wiki is a reference guide. It doesn't replace the need for a new version of the Asterisk handbook. One thing the handbook covers better is Zaptel/digium hardware, the Wiki is weak on that subject. Propably because I haven't got any Zaptel hardware and the handbook is sufficient for other writers, so they haven't added that subject either. Please give me some more clues on how the Wiki takes you in the wrong direction a lot. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP URIs and ENUM or other types of lookup
But that would sort of break SIP. A SIP URI is [EMAIL PROTECTED], so it makes No, A SIP URI is [EMAIL PROTECTED] - there's a big difference. Read on DNS SRV records on http://www.voip-info.org/tiki-index.php?page=DNS%20SRV Quoting myself: No one really mails [EMAIL PROTECTED] any more. We're mailing [EMAIL PROTECTED] and the DNS MX records helps the mail client to send the mail to the correct mail server. Why should we call [EMAIL PROTECTED] instead of using [EMAIL PROTECTED] Remember that the later construction in addition to being user friendly, also adds redundancy, and with a proper DNS configuration also may add load balancing. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tuning the Linux kernel?
Olle, are you watching, this is for the Wiki. I'm here, trying to catch up :-) Don't forget that applications are also modules and can be set to not load. I don't list applications here as they have been listed elsewhere. http://www.voip-info.org/tiki-index.php?page=Asterisk+modules Thank you! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
On Thu, 20 Nov 2003 08:44:10 -0700, Chris Hirsch wrote Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com - L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. As somebody already pointed out (and I corrected this in the Wiki a few days ago), IAX uses UDP, so plain ssh tunneling won't work. Anybody have any ideas? I did notice that when I start gnophone I see iax.c line 654 in iax_init: Started on port 5036 Listening on port 5036 and it doesn't seem to matter what I do inside the config. Are these ports in some way hardcoded? If if they are can't I do something like above? Thanks! Chris The 5036 port is hardcoded in the IAX library (iax.h, #define IAX_DEFAULT_PORTNO 5036) which gnophone uses. For the quickest hack, change the value and recompile libiax. In the longer run, consider adding a command line option (or even better, a GUI config item), specifying an int passed to iax_init() in pc_init() in phonecore.c in gnophone source (I went through the code so much I almost know it by heart ;). Currently it is passed a zero which means the default. In any case, if the required port is in use, IAX uses a random one. Grzegorz Nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
John Todd wrote: At 11:47 AM -0600 11/20/03, Tilghman Lesher wrote: (SIP, Zap, whatever) has their own CLASS dialplan sets, then that is a different problem - either deactivate them and use the server, or leave them enabled and ignore things for that line. How exactly did you hack your dialplan so that forwarding works? Obviously, it can't be transparently handled without inserting code in the channel allocation routine. Or are you talking about simple forwarding, like Call Forward Unconditional? That's _really_ easy... just a DBGet (or external DB lookup) call. http://www.voip-info.org/wiki-Asterisk+call+forwarding /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Opteron - Kernel optimizations
I am considering buying a Quad Opteron for asterisk, However i'd like not to buy one and see that it aint working ;) - would asterisk compile for opteron ? - would ilbc compile for opteron ? - would the g729 license work on opteron ? - would zaptel compile and run ? (TE410p) And if not, would it demand a lot of work to get it running ? Would using 64 bit actually be faster for decoding encoding ? (i mainly use iLBC, googled around but found nothing on x64 optimizations) Any one with hands on experience on Asterisk vs Opteron ? Did anyone so far bother to compile asterisk on the intel c compiler ? Any speed gains ? I'm also very interested in anything that could give me a speed gain, compiler settings, kernel tweaking, etc etc... let me know :) zoa. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 without iaxtel.com
Greetings everyone. Could anyone tell me how to setup an IAX call using iaxcomm from a remote (PC) user without going throug iaxtel.com? I would like users to register to my server directly instead of looking up in iaxtel directory. Please provide an example of iax.conf commands and extensions.conf. Your help would be greatly appreciated. Thanks in advance. Ricky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tunnel iax via gnophone with ssh?
Grzegorz Nosek wrote: On Thu, 20 Nov 2003 08:44:10 -0700, Chris Hirsch wrote Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com - L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT. As somebody already pointed out (and I corrected this in the Wiki a few days ago), IAX uses UDP, so plain ssh tunneling won't work. if you need a secure tunnel use vtun - it's just a grate software for building secure tunnels :))) http://vtun.sourceforge.net/ We are using it without any problem for VoIP Lubo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgrade CISCO 7960 Question
Quoting Walker Haddock [EMAIL PROTECTED]:
Thanks for answer.
I already did it and it is working fine.
Bart
On Fri, Sep 24, 2004 at 07:13:39AM -0400, [EMAIL PROTECTED] wrote:
Yes, it is. But why would you want to do that when yo said what you
want it to be at 6.0.
He's got the Skinny version and wants to change to the SIP version.
Maybe you didn't expling what and why you want to do it in enough detail
to get a good answer.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Jozwiak
Sent: Friday, November 21, 2003 8:11 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Upgrade CISCO 7960 Question
Hello,
My Cisco phone has software:
Boot Load: PC030300
Ver: 3.2(7.0)
The SIP version 6.0 has these versions if you login to the 7960 tusing
telnet:
Cisco Systems, Inc. Copyright 2000-2003
Cisco IP phone MAC: 0030:94c2:ea67
Loadid: SW: P0S3-06-0-00 ARM: PAS3ARM1 Boot: PC03M030 DSP: PS03AT38
And I want to upgrade it to SIP 6.0
Is it possible or I have to upgrade to ealier then 6.0 and then
to 6.0 ?
I upgraded directly from your version of the Skinny image to the SIP 6.0
image and it is working fine.
bart
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
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[Asterisk-Users] Experimental Switzerland - IAX gateway
Hi, to test my Asterisk / IAX connection I have configured the Swiss phone number 032 841 47 74 to a IAX gateway. You can dial 1-700, 1-800 and other numbers from this number (prefix with 00: for example 0018005551212). This is a local rate number. I have not yet implemented IAXtel - Swiss 1-800 yet because I didn't succeed in registering two IAXtel numbers yet. Feel free to test this, for example to test dialing into your gnomephone application. It may be stopped at any time, will probably work mostly week-end and working hours GMT+1 at this time. This test might however be discontinued at any time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
Steven Sokol wrote: 1. Redial 2. Voicemail Box Monitoring 3. Enhanced Conferencing 4. Outlook/Act/Goldmine Integration (PIM stuff) 5. Call History (both inbound and outbound) 6. Redirect Option on Ring (VM, Application, Transfer, etc.) 7. Automatic mixing and delivery of monitored (recorded) files. What would be neat would be a limited setup for secretaries or receptionists to be able to see incoming calls on all extensions and be able to forward them to thier phones... like a DSS. Maybe there's a way to do it now, but not that I know of. A copy of the source code (let's call this LGPL for now) is available here: http://www.sokol-associates.com/Downloads/AstMgr.zip It's written in VB6 (yes - barf, gag, whatever). The only thing required beyond the integral VB6 controls is the Windows Scripting Runtime which most PCs should have. I will work on an installable version soon. I may also port it to something more cross-platform. Please bear with me as I am just learning Gnome/GTK/X-windows. May I suggest you use something like HBasic -- it's like Visual Basic, but can be used on Windows and Linux. Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
I'm receiving calls on my asterisk server from iconnecthere. My asterisk server is behind nat but it still seems to be working fine. AJ On Fri, 21 Nov 2003, Chris HARIGA wrote: Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 without iaxtel.com
Hi, Ricky On Sat, 22 Nov 2003 03:15:27 -0800, Asterisk [EMAIL PROTECTED] wrote: Greetings everyone. Could anyone tell me how to setup an IAX call using iaxcomm from a remote (PC) user without going throug iaxtel.com? If you want to call PC-toPC, just type 192.168.0.1/s just above the Dial key. No need to register with iaxtel.com I would like users to register to my server directly instead of looking up in iaxtel directory. Please provide an example of iax.conf commands and extensions.conf. My laptop registers with my home asterisk server, vangate, as extension 309 extensions.conf: exten = 309,1,Dial(IAX2/309) iax.conf: [309] type=friend host=dynamic secret=oops_I_forgot _to_change_this context=from-iax callerid=Michael PC 309 Hope this helps. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Are you also able to make outgoing calls via Iconnecthere? If so do you mind posting your config? I tried their 10 minute trial a couple of months ago but was not able to get a connection. Thanks, Robert I'm receiving calls on my asterisk server from iconnecthere. My asterisk server is behind nat but it still seems to be working fine. AJ On Fri, 21 Nov 2003, Chris HARIGA wrote: Hi, Is anyone using the iconnect on Asterisk to receive and to place calls? Best regards, Chris HARIGA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
1. Redial 2. Voicemail Box Monitoring 3. Enhanced Conferencing 4. Outlook/Act/Goldmine Integration (PIM stuff) 5. Call History (both inbound and outbound) 6. Redirect Option on Ring (VM, Application, Transfer, etc.) 7. Automatic mixing and delivery of monitored (recorded) files. What would be neat would be a limited setup for secretaries or receptionists to be able to see incoming calls on all extensions and be able to forward them to thier phones... like a DSS. Maybe there's a way to do it now, but not that I know of. I have looked at creating a Console version of the application. It would be very much like a DSS (Direct Station Selector for the non-ATT/Avaya initiated). It would support either click-to-transfer or drag-and-drop transfer of incoming calls. May I suggest you use something like HBasic -- it's like Visual Basic, but can be used on Windows and Linux. HBasic? Cool. I have had somebody else suggest wxWindows as a method of building the GUI. Do the two play nicely together? Steve Sokol Sokol Associates, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
For the benefit of others who may experience this (multiple frame rejections and PRI read errors under high IVR call volume- E1 circuits) I've discussed this with Mark at Digium, and he's called into my system. There may be a problem with PRI software frame buffering with a high volume of call setups. I will create a bug report so that this can be tracked. In the meantime, if possible I'd like to verify this problem on another system. If anyone has a TE410P and would like to try my load tester on their system, could you please contact me *off-list*, and I'll send you my call generation Perl script for you to try. You would need one E1 crossover cable: (This is simple to construct from a CAT5 Ethernet patch cable). I can make the problem occur with only 30 sending and receiving channels on the same system... THANKS! Thanks to Mark and Martin for their help on this. Scott M. Stingel Emerging Voice Technology Inc. [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Friday, November 21, 2003 9:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P Errors under load Again, can you please confirm you are neither running serial console *nor* graphical console (e.g. framebuffer). If you can call into the office we can ssh in and take a look at the configuration. Mark On Fri, 21 Nov 2003, Scott Stingel wrote: (Apologies: starting this as a new thread - I'm in a new location.) Mark- Ran latest CVS from today, and sorry to report little improvement with the changes you made. Running my IVR load test from one span to another on same system. I'm initiating calls on the 2nd span, these are channels 32-62 (skipping the D channel 47), and receiving on the cooresponding channels on the 1st span, channels 1-31 (D channel is 16). When I run these 30 channels, I get hundreds of WARNING's (excerpt below). I'm using a short crossover cable (1,2 = 4,5) When I run only 10-15 channels, I get few or no WARNING's... Note read error on channel 252(?) Why is asterisk retransmitting so many frames on each error? These symptoms are identical to those that I've been getting from my customer in the field, while connected to a DMS-100, handling real traffic. THANKS Scott WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 253 failed: Unknown error 500 WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 253 failed: Unknown error 500 WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 97 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 98 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 99 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 100 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 101 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 102 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 103 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 104 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 105 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 106 now, updating n_r! WARNING[1175660608]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: !! Got reject for frame 97, retransmitting frame 107 now, updating n_r! WARNING[1167272128]: File chan_zap.c, Line 5716 (zt_pri_error): PRI: Read on 252 failed: Unknown error 500
RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
Zoa,
When the boxes are red, that usually indicates that the channel is busy.
In the screen shots I sent earlier you will see that one of the buttons
is red:
http://www.sokol-associates.com/images/AstMgr.jpg
Notice that only the station marked Test Xten is red. This station is
busy (on another call). I don't know if that has anything to do with
your issue, but I thought I would throw that out.
The message you reference below is a Status message. In this program
the Status messages really only serve as keep-alives. Every 30 seconds
the system issues a command Action:Status to keep NATs from closing
the connection due to lack of traffic.
Try this: open the command window and try manually executing some of
the CLI commands. Try sip show peers to make sure the SIP peers are
registered. Also try sip show channels to see if there is already a
call terminated at the channel you are calling.
I will try to diagnose this further if you can send some additional
information. Please include the monitor.conf file, and if possible a
- trace from Asterisk.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of zoa
Sent: Saturday, November 22, 2003 4:13 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1
(Alpha)
[201]
Username=davy
Technology=SIP
DeviceID=davy
[202]
Username=pieter
Technology=SIP
DeviceID=pieter
201 is the extension from in extensions.conf
davy = the thing between brackets in sip.conf
When i try to click on one of the red boxes in the manager, i always
get:
Event: Status -|- Channel: SIP/davy-1a07 -|- CallerID: davy -|- State:
Up
-|- Context: sip -|- Extension: 202 -|- Priority: 1 -|- Link:
SIP/pieter-e582 -|- Uniqueid: 1069581.102
Response: Error -|- Message: Invalid channel
At 21:06 21/11/2003 -0600, you wrote:
Here's the structure for the monitor.conf file:
[1101] Extension Number (from extensions.conf in
Asterisk)
UserName=Blah Blah Label. Simply sets the caption for the
button.
Technology=SIP Technology used for stations (SIP, MGCP, Zap,
etc.)
DeviceID=1101 Device identifier (from sip.conf in this
case)
All of the Technology values are normal asterisk values except for
APP,
which is an application (like Voicemail or MOH or MeetMe) and PSTN,
which is a number outside of the Asterisk inside dial plan.
I hope this helps. Remember that for PSTN and APP values, the
bracketed
Extension number and the DeviceID need to be the same.
Regards,
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of zoa
Sent: Friday, November 21, 2003 7:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Call Manager for Windows
0.0.1
(Alpha)
Could you give me some explanation on how to use the configuration
file ?
I always get INVALID channels if i click on the red icon next to
my
name.
(Maybe i should use a numeric context or use numeric user names?)
Tool looks great, this will be a very cool asterisk addition.
zoa.
At 16:43 21/11/2003 -0600, you wrote:
I think the script host gets installed with Windows explorer. If
you
don't have it, you can use the DLL in the dlls download:
http://www.sokol-associates.com/Downloads/Dlls.zip
Hope that helps.
Thanks,
Steve
-Original Message-
From: [EMAIL PROTECTED]
[mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Walker Haddock
Sent: Friday, November 21, 2003 4:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk Call Manager for
Windows
0.0.1
(Alpha)
here:
http://www.sokol-associates.com/Downloads/AstMgr.zip
It's written in VB6 (yes - barf, gag, whatever). The only
thing
required beyond the integral VB6 controls is the Windows
Scripting
Runtime which most PCs should have. I will work on an
installable
version soon. I may also port it to something more
cross-platform.
Please bear with me as I am just learning
Gnome/GTK/X-windows.
Steve, how do you know if the Windows Scripting Runtime is
installed
in
Windows XP Pro?
Where do you get it from and how should it be installed?
Thanks, Walker
--
DataCrest, Inc. -- Technically Superior
**
Walker Haddock http://www.datacrest.com
DataCrest, Inc.e-mail: [EMAIL PROTECTED]
1634A Montgomery Hwy.phone: 1-888-941-3282,
1-205-335-8589
Birmingham, AL 35216 fax: 1-205-823-7838
***
___
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[Asterisk-Users] How to dial out using OH323?
Hi I am trying to dial an extention on my gateway using OH323 without a gatekeeper. I would like to be able to do this: exten=_8.,1Dial(OH323/($EXTEN:1)@xxx.xxx.xxx.xxx,20,r) It seems that the only way I can dial via OH323 is exten=_8.,1Dial(OH323/xxx.xxx.xxx.xxx,20,r) Any incite into diling with OH323 will be appreciated. Thank you, Serge _ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=dept/bcommpgmarket=en-caRU=http%3a%2f%2fjoin.msn.com%2f%3fpage%3dmisc%2fspecialoffers%26pgmarket%3den-ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P configuration Problem
Hi People, I have the following scenario: PSTN via Ibercom - 3 x X100P - Asterisk - Sip phones Ibercom = A product of Telefonica in Spain, interconnecting with old Ericsson equipment buildings of the same company via PRI and also connecting with PSTN via PRI. My problem is that when I have an entry call via X100P and I redirect this call to the voicemail or conference room. The caller give the msg and when hang up the voice mail save 180s of busy tone until timeout and hangup the zap channel or i see the busy tone in conference room until the call timeout. If i answer the call in the Sip phones when I hangup the Zap channel also hangup correctly. I think that I have correctly the indications.conf. Someone have any similar issue or know some workaround? [es] description = Spain ringcadance = 1500,3000 dial = 425 busy = 425/250,0/250 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 regards, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list configuration issues...
That said, I find an FAQ quite a good idea. Maybe just as another page on the voip-info.org Wiki? http://www.voip-info.org/wiki-Asterisk+FAQ It's been there for a while now. Thank you, anyhow, for suggesting improvements. /O ;-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap MWI method
What method does the Zap MWI use? FSK, 48 volt, or 90 volt? --Eric -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. call generation Perl script for you to try. You would need one E1 crossover cable: (This is simple to construct from a CAT5 Ethernet patch cable). I can make the problem occur with only 30 sending and receiving channels on the same system... THANKS! -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - phone docs
Rich Adamsson and I have started a new Wiki page to document configuration for different VoIP clients - both hardware and software. http://www.voip-info.org/tiki-index.php?page=Asterisk%20phones Rich started with writing documentation on the Cisco 79xx phones. Please help us adding information for other phones! Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Posted At: Friday, November 21, 2003 10:17 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Friday, November 21, 2003 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
Since you say ADSI I assume you mean analog on a Zap channel. Send a FLASH on the line and MOH will start. On Sat, 2003-11-22 at 12:59, PBX wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Posted At: Friday, November 21, 2003 10:17 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Friday, November 21, 2003 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
At 10:59 AM 11/22/2003, you wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? What kind of phone do you have? MOH depends first on the phone, as it is the phone that decides what to do when you press the hold button. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
But the problem with that is the user then hears dial tone. And if they hang up the line it rings them back... The only way I have been able to get anything like what I want is, to push flash then the hold button... That is not the exact motion I want a user to go through. If there is a way to signal the hold button in ADSI that would work.. Cause then I could request a flash then hold sequence and then to take them off hold, I could just do the opposite sequence. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Posted At: Saturday, November 22, 2003 2:07 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Since you say ADSI I assume you mean analog on a Zap channel. Send a FLASH on the line and MOH will start. On Sat, 2003-11-22 at 12:59, PBX wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Posted At: Friday, November 21, 2003 10:17 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Friday, November 21, 2003 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
On Sat, 22 Nov 2003, PBX wrote: But the problem with that is the user then hears dial tone. And if they hang up the line it rings them back... The only way I have been able to get anything like what I want is, to push flash then the hold button... That is not the exact motion I want a user to go through. If there is a way to signal the hold button in ADSI that would work.. Cause then I could request a flash then hold sequence and then to take them off hold, I could just do the opposite sequence. Try call parking... -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP URIs and ENUM or other types of lookup
On Sat, Nov 22, 2003 at 11:28:34AM +0100, Olle E. Johansson wrote:
Quoting myself:
No one really mails [EMAIL PROTECTED] any more. We're mailing [EMAIL PROTECTED] and
the DNS MX records helps the mail client to send the mail to the correct
mail server. Why should we call [EMAIL PROTECTED] instead of using [EMAIL PROTECTED]
Sure, no question. But that wasn't really the point.
The problem was, how, in extensions.conf to deal with Dial strings
of the form IAX/iaxpeer/number and Zap/n/number and SIP/[EMAIL PROTECTED]
in a consistent way -- i.e. ${FOO}/${EXTEN} does not work with SIP
so you would have to special-case any Dial strings that want to use
this form...
Another example: first priority, try to dial out the PSTN, second
priority, dial some SIP gateway:
PRIMARY=Zap/0
BACKUP=SIP/someproxy
exten = _.,1,ChanIsAvail(${PRIMARY}${BACKUP})
exten = _.,2,Dial(${AVAILCHAN}/${EXTEN})
it would be nice if this sort of thing were possible to do this...
-w
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RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
As someone else said, that's really a function of the phone. As far as I know there is no ADSI command for HOLD or Start MOH. On Sat, 2003-11-22 at 13:15, PBX wrote: But the problem with that is the user then hears dial tone. And if they hang up the line it rings them back... The only way I have been able to get anything like what I want is, to push flash then the hold button... That is not the exact motion I want a user to go through. If there is a way to signal the hold button in ADSI that would work.. Cause then I could request a flash then hold sequence and then to take them off hold, I could just do the opposite sequence. -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Posted At: Saturday, November 22, 2003 2:07 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Since you say ADSI I assume you mean analog on a Zap channel. Send a FLASH on the line and MOH will start. On Sat, 2003-11-22 at 12:59, PBX wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Posted At: Friday, November 21, 2003 10:17 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Friday, November 21, 2003 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
Yes it is different, the E1 crossover cable pairs 1+2 with 4+5. What I meant was that you can make a crossover by cutting up a CAT5 cable and redoing two pair (I wan't clear) Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Burkholder Sent: Saturday, November 22, 2003 6:43 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TE410P Errors under load Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. call generation Perl script for you to try. You would need one E1 crossover cable: (This is simple to construct from a CAT5 Ethernet patch cable). I can make the problem occur with only 30 sending and receiving channels on the same system... THANKS! -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP channel improvements
I just discovered that the SIP channel has undergone some major improvements. I'm now able to dial any SIP URL with dial, couldn't get it to work earlier, all domains had to be defined in SIP.conf. This, in addition to the SIPDOMAIN variable, makes the SIP channel even more useful. Thank you, Mark, for your additions! Now, ENUM/E.164 will propably work even better. I'll give it a try. Now, to be the documentation-pain-in-the-*** I would like to get an explanation of the autocreatepeer SIP.conf setting and functionality? It's not in sip.conf.sample yet. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Saturday, November 22, 2003 1:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? -gcc There is a seperate module for the phones I use that will do Music on hold for all of the phones in the group. These are not ADSI phones, I guess you might call them analog feature phones. I don't think that Flash is a very good way to effect hold, since it is also used for other functions, so I guess the answer is that it probably won't work like you want it to without some changes. HTH Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Posted At: Friday, November 21, 2003 10:17 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of PBX Sent: Friday, November 21, 2003 6:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Ok... I know I have asked this question before, but have never gotten an answer... When I press the hold button on my phone, should the caller hear music just like when I park the caller or transfer them to another extension? It depends... If the button uses a function of the phone to hold the call(ie keeps the call active but mutes the speaker mic),then you will not hear music on hold from *. I have some phones that can operate this way. Andy Please assist... -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P configuration Problem
On Sat, Nov 22, 2003 at 06:05:06PM +0100, Daniel Concepcion wrote: My problem is that when I have an entry call via X100P and I redirect this call to the voicemail or conference room. The caller give the msg and when hang up the voice mail save 180s of busy tone until timeout and hangup the zap channel or i see the busy tone in conference room until the call timeout. I have seen similar things. I believe it has to do with the difficulty in detecting when the far end has hung up in the absence of digital signalling information. With analogue interfaces like the X100P usually they will detect a reversal in polarity and take that to mean that the far end has hung up, and then hang up themselves. But this is with telco provisioned POTS loops. I suspect your Ibercom box is not reversing the polarity, so the X100P has no way to tell that the call is gone. Perhaps it is possible to configure the Ibercom to make it do the right thing? -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP channel improvements
On Sat, Nov 22, 2003 at 08:51:35PM +0100, Olle E. Johansson wrote: I just discovered that the SIP channel has undergone some major improvements. But not, alas, in the realm of NAT. Is there any possibility of removing the broken externip implementation and importing the patch I submitted that does it properly? If there are objections to the patch, please say so and I will attempt to deal with them. Maintaining a forked chan_sip.c and importing the other changes is beginning to be a bit of a headache and making it hard to keep up with CVS... -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: tunnel iax via gnophone with ssh?
On Fri, Nov 21, 2003 at 12:52:24PM +0200, Michael Manousos wrote: No, they don't. H.323 uses TCP. And SIP has an option to use TCP No, H.323 uses UDP for voice (RTP). It doesn't make sense to use TCP for voice transport. And likewise for SIP I believe, it will just use TCP for signalling. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New DIAX - version 0.9.4 - a big step forward - available for download
Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate file; - can configure your own callerID (Name and Number); - no need to re-enter credentials when entering the registration window; - the application is even smaller - less than 100K for all the files (except help file); - ordering calls in lists (last on the first position by default, click to order after columns); - double-click on missed calls indicator to open the missed calls list; - add ',' as key for '#'; - ESC key can be used to exit from any secondary window (Cancel button equivalent); - display the type of the message in the status bar (STATUS/NOTICE/ERROR/IAXMSG); - ask for audio settings at first run; - tab key in Registration window select the full field; - solved bug: crash at exit; - solved bug: phonebook open in the back when main form always on top if calls list opened first; - solved bug: app crash when the number string starts with the name; - solved bug: cannot dial any entry from the calls list (just first number from the list is dialed) Please send me your feedback. Best regards, Dan : ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g729 codec questions error running asterisk now
Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try again... Cannot allocate channels... Process Stopped! Error -10 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 Segmentation fault (core dumped) Here is /var/log/asterisk/messages for this Nov 22 12:25:36 WARNING[1167272000]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable Nov 22 12:25:36 WARNING[1074420448]: File chan_zap.c, Line 7203 (load_module): Ignoring rxwink Nov 22 12:25:37 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 I didnt have a problem before. Kind Regards, Steven Kalcevich MSN:[EMAIL PROTECTED]
[Asterisk-Users] Re: SIP channel improvements
Olle E. Johansson wrote: I just discovered that the SIP channel has undergone some major improvements. I'm now able to dial any SIP URL with dial, couldn't get it to work earlier, all domains had to be defined in SIP.conf. ...and I'm able to call any SIP URL with Xlite, with Asterisk resolving the domain part according to DNS SRV records, contacting the right SIP proxy for the DOMAIN, setting up the call. This is brilliant, a major step forward for the SIP support in Asterisk! /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
Jared Smith wrote: Leif Madson and I are (slowly) working on getting something in place to do just that... Feel free to join us in in #asterisk-doc to hammer out details, ideas, etc. Agreed. I am home for the weekend now, so I should be getting some stuff hammered out. Will be reviewing what Sokol has written, as well as all the work Jared is doing on getting the backend stuff setup (DocBook, CVS, etc.. etc..). Basically, I started the initiative, but Jared is doing all the work right now! :) -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ |@| leif at hacklocalhost dot com | | SMS| sms at hacklocalhost dot com | | FWD| 18924 IAX| 1-700-363-0761 | |iptel| 8972-1969sipph| 1-747-386-1618 | +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
A 568B Ethernet cable will be paired as follows: 1 ORG-WHT 2 ORG 3 BLU 4 GRN-WHT 5 GRN 6 BLU-WHT 7 BRN-WHT 8 BRN As you can see you still maintain twisted pair integrity for T1 applications (1-2, 4-5), as well as Ethernet (1-2, 3-6). The primary issue would be shielding as most Ethernet cables aren't shielded. THX/BDH On Sat, 2003-11-22 at 13:43, Ray Burkholder wrote: Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stability with the Supura SIP Units
Hi, I would like to know if those using the Sipura SIP units with Asterisk have found them to be stable. I ask because the Grandstream units simply have not improved their stability considerably, and we are now in search of an alternate to the ATA186. We want to know if anybody has seen the sipura unit stop registering (like what happens with the Grandstream phones). We have ordered some sipura units for lab testing but would also like some feedback from those on the list. Feel free to answer directly if you feel this discussion is not appropriate here. Thanks, Andres
[Asterisk-Users] Help Required for Speex
Hi Have any body experienced Asterisk with Speex??May i know the result i.e Voice quality or echo problems and wats frame size and other settings are compataible with asterisk . Regards Oabid _ The new MSN 8: advanced junk mail protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iconnect (DeltaThree) config on Asterisk
Actually I'm only using it for incoming calls; however I believe John Todd has sample configs posted on his site and these should include some examples for iconnecthere. His site is http://www.loligo.com/asterisk. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
Yes, that's true how the pairs are formed, and its also true that a (short) ethernet cable can normally be used as a straight through E1/T1 cable, since all wires are connected. We need an E1 crossover cable for the tests we're running, and so we have to make one, because an ethernet crossover cable will not work. For anyone trying to make an E1/T1 crossover, here's a nice diagram from NMS that may help. The only pins that are needed in a short cable for testing are 1,2,4,5. Click on: http://www.nmscommunications.com/NMS/nms_technotes.nsf/0/91d49c8785b2aab0852 566fa0050740a?OpenDocument Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian D Heaton Sent: Saturday, November 22, 2003 10:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TE410P Errors under load A 568B Ethernet cable will be paired as follows: 1 ORG-WHT 2 ORG 3 BLU 4 GRN-WHT 5 GRN 6 BLU-WHT 7 BRN-WHT 8 BRN As you can see you still maintain twisted pair integrity for T1 applications (1-2, 4-5), as well as Ethernet (1-2, 3-6). The primary issue would be shielding as most Ethernet cables aren't shielded. THX/BDH On Sat, 2003-11-22 at 13:43, Ray Burkholder wrote: Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 codec questions error running asterisk now
run ldconfig, reboot the server and normally all will be fine, if not you will have to reregister. I've seen it before, and i'm sure you will see it again :-p At 16:20 22/11/2003 -0500, you wrote: Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try again... Cannot allocate channels... Process Stopped! Error -10 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 Segmentation fault (core dumped) Here is /var/log/asterisk/messages for this Nov 22 12:25:36 WARNING[1167272000]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable Nov 22 12:25:36 WARNING[1074420448]: File chan_zap.c, Line 7203 (load_module): Ignoring rxwink Nov 22 12:25:37 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to initialize va stuff: -1 I didnt have a problem before. Kind Regards, Steven Kalcevich MSN: mailto:[EMAIL PROTECTED][EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy
Aastra 350 -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Posted At: Saturday, November 22, 2003 2:08 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy Subject: RE: [Asterisk-Users] MOH - Hold Button - I think I'm going crazy At 10:59 AM 11/22/2003, you wrote: Is there a solution to have the hold button to play MOH. Or even some type of ADSI function that allows for this? What kind of phone do you have? MOH depends first on the phone, as it is the phone that decides what to do when you press the hold button. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)
Steven Sokol wrote: I have looked at creating a Console version of the application. It would be very much like a DSS (Direct Station Selector for the non-ATT/Avaya initiated). It would support either click-to-transfer or drag-and-drop transfer of incoming calls. Excellent! This is one feature we really need! The way I thought about doing this would be to stick the console as an icon in the System Tray (where the volume control, et. al. is) and have it pop up a window (similar to Gnome's calender/date applet) whenever a call comes in that must be brought to the person's attention. For example, if Bob's secretary wasn't answering her phone (or had marked herself as away), the console would pop up for all the other secretaries with the incoming call. Since they'd know the extension, it would be transparent to the caller that it was really Jane's secretary answering. May I suggest you use something like HBasic -- it's like Visual Basic, but can be used on Windows and Linux. HBasic? Cool. I have had somebody else suggest wxWindows as a method of building the GUI. Do the two play nicely together? Well, HBasic uses Qt for its GUI and wxWindows is a GUI library for C/C++ (and Perl, and...) similar to Qt. So basically, they're apples and oranges. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * on 2.4.20-gentoo-r8 linux with eicon-diva-pro-pci-2.0 isdn-bri card how-to ???
hi, sorry my english :( how-to configure 2.4.20-gentoo-r8 linuxand * for use aditional eicon diva pro pci 2.0 isdn-bri card ??? planing to use isdn card for voice access to PSTN i´m very happy with *, now using one X100P, but have new isdn-bri connection and like to route my 3 phone lines to sip phones at ofice in working hours. never use ISDN card before and no free pci slots for 3 X100P solution please help thanks, Carlos Madrid
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thu, Nov 20, 2003 at 02:52:04PM -0700, Jared Smith wrote: Rather than telling newbies (especially the technically challenged) to google for it, we could send them a link to the ebook and tell them to run the search in Acrobat reader to find the answer. Anybody want to start a thread building the outline for the book? Leif Madson and I are (slowly) working on getting something in place to do just that... Feel free to join us in in #asterisk-doc to hammer out details, ideas, etc. please please please if you are going to write something like that, write it using something like texinfo or groff or docbook or whatever so that you can make it available in a wide variety of formats. you should not have to run non-free software in order to be able to search the documentation (and no, acroread is not free software) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAQ, Documentation, How-to, etc
On Thu, Nov 20, 2003 at 04:55:11PM -0600, Steven Sokol wrote: Does anybody want to help with this? PLEASE EXPAND MY OUTLINE. ADD THINGS YOU THINK ARE IMPORTANT. LET ME KNOW IF YOU WANT TO TAKE A CRACK AT A CHAPTER! This book will be available in electronic form under some sort of open publishing license, in addition to being sold in bookstores, right? -w ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P Errors under load
One of the simplest ways to make a T-1 (E-1) crossover is to take a dual RJ-45 biscuit and do the crossover inside it. Label it as a crossover with a Sharpie. Then you can use two normal cables. I always had at least one of these in my kit when I was doing field work. THX/BDH On Sat, 2003-11-22 at 19:45, Scott Stingel wrote: Yes, that's true how the pairs are formed, and its also true that a (short) ethernet cable can normally be used as a straight through E1/T1 cable, since all wires are connected. We need an E1 crossover cable for the tests we're running, and so we have to make one, because an ethernet crossover cable will not work. For anyone trying to make an E1/T1 crossover, here's a nice diagram from NMS that may help. The only pins that are needed in a short cable for testing are 1,2,4,5. Click on: http://www.nmscommunications.com/NMS/nms_technotes.nsf/0/91d49c8785b2aab0852 566fa0050740a?OpenDocument Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian D Heaton Sent: Saturday, November 22, 2003 10:04 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] TE410P Errors under load A 568B Ethernet cable will be paired as follows: 1 ORG-WHT 2 ORG 3 BLU 4 GRN-WHT 5 GRN 6 BLU-WHT 7 BRN-WHT 8 BRN As you can see you still maintain twisted pair integrity for T1 applications (1-2, 4-5), as well as Ethernet (1-2, 3-6). The primary issue would be shielding as most Ethernet cables aren't shielded. THX/BDH On Sat, 2003-11-22 at 13:43, Ray Burkholder wrote: Might you be getting problems because you are using an Ethernet cable? If my memory serves correctly, an Ethernet cable is paired differently than an E1/T1 cable. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bayonne and Asterisk
How about issues such as echo, voice quality, supported codec's? does it work with SIP? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Tuesday, November 18, 2003 9:46 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Bayonne and Asterisk I used Bayonne for 2 years before switching to Asterisk. Right now I'm still running Bayonne on one application and it's been running happily without me looking at it for over 6 months. I'd say these are the strengths of Bayonne: - Runs on Dialogic, Pika and other widely available hardware - extremely reliable, mine never crashes and here are the weaknesses: - nowhere near as active of a support community as Asterisk has - configuration of the hardware/drivers is a nightmare compared to Asterisk/Digium - it is quite limited in it's included apps, IVR and voicemail - not as many options for scripting as Asterisk - it was not designed to have full PBX functionality, some PBX functionality is added as afterthought - the code/organization/flow is not as well thought out or documented as Asterisk is And yes, they can run fine together(I'm not using VOIP, just a T1 out of Asterisk to Bayonne to test and see if it would work). The IVR application that I currently still have running on Bayonne is only still on Bayonne because it can never go down, and Bayonne has proven itself to me to be extremely stable, while I cannot personally say AT THIS TIME that an Asterisk box would stay up for over 6 months with no crashes. At last check I was never able to get VOIP inbound working on Bayonne, maybe this has changed in the last 6 months but if you do get it working I'd be interested to find out how. MATT--- -Original Message- From: Dirk-Jan Wemmers [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 8:44 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bayonne and Asterisk All, is anyone using Bayonne in conjunction with Asterisk? I'm currently using only Bayonne, but I'm investigating the possibilities of switching the telephony frontend over to Asterisk, and have Asterisk route the IVR tasks to Bayonne through H323. Anyone care to share his views on this approach? Any pointers or do's and don'ts? All info is greatly appreciated! Regards, Dirk-Jan -- Dirk-Jan Wemmers, Capcave B.V. Zonnebaan 17, 3542EA Utrecht T +31(0)30-2149670, F +31(0)30-2149679 M +31(0)651 063040, E [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Local numbers to Victorville/Apple Valley, CA
Hey all, I am in the High Desert region of southern California, USA. I was wondering if any of the SIP providers offer numbers serviced out of the following Verizon central offices: Apple Valley (Apple Valley CO/APVYCAXF) Apple Valley (Desert Knolls CO/DSKNCAXF) Victorville (VTVLCAXA) Adelanto (ADLNCAXF) Hesperia (HSPRCAXF) These are the COs which offer prefixes which are local calls from my home. I don't believe there are any facilities-based LECs in this area, which is why I asked about providers colo'ing at the COs listed above, but if there are any facilities-based providers (they'd probably be in Victorville) I'd be willing to hear about them too. Thanks in advance. -- JustThe.net Internet New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
