RE: [Asterisk-Users] No Ringing from PSTN
Well, the ATA uses SIP to communicate with the * box. SIP by default doesn't generate a ringing indicator when the far side is ringing, you indeed DO have to tell it to ring, using the r flag in the extension. **Note, this is just from my experience. -- Josh Roberson Indigent Networks 1.877.677.9647 x1 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dolloff Sent: Thursday, October 09, 2003 3:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] No Ringing from PSTN That does make a ringing sound, but any idea what's causing the problem? Stephen Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.522 / Virus Database: 320 - Release Date: 9/29/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
TeleSIP wrote: Mark's words to me, when I was a newbie: [00:08] kram a user is to authenticate an incoming call [00:08] kram a peer is someone you send a call to [00:08] kram friend, of course, is both I am still at a loss here. If both are set to peer then how can either end originate the call? You would need at least one end to be user or friend. Have both a user and a peer, they can be the same name. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5 second latency sip to oh323
hi michael, transferred the call using callmanager. the weird thing is, the unusual latency is present only in x-lite and not present in windoze messenger. i played around and changed the codecs but didn't find anything unusual... the root of the problem lies in the call hold, when i place a call on hold, upon resume, the audio becomes lagged when using messenger, the audio initially is lagged but is able to catch up. a series similar to this line comes up on console upon resume I/O 78216400160 320 Late4 4 14.632 0.020 14.715 0.020 0.083 - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 9:25 PM Subject: Re: [Asterisk-Users] 5 second latency sip to oh323 How do you transfer the call? Michael. Kelvin Chua wrote: hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred the scenario is this: sip-asterisk-h323:operator (who then transfers the call) h323:destination --audio path 5-second latency audio path ok--- here is the output of the show channels H323:19742 (voip s1 ) Up Bridged Call SIP/kelvin-6952 SIP/kelvin-6952 (voip 2010 1 ) Up Dial OH323/H323:[EMAIL PROTECTED]|25|mt the problem only exists in transferred calls any info would be appreciated thanks =) ~kelvin ___ 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] Howto configure asterisk with AVM ISDN card
Hi guys, I'trying to setup Asterisk with an AVM Fritz PCI ISDN card with no success. I have an IP phone from Micronet and I want to receive and place calls over ISDN. But I'm not sure if I need the binary CAPI from AVM or isdn4linux. I've tryed with redhat 9 and with Suse 8.2. Cold someone help me? Thank you very much. _ The new MSN 8: smart spam 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] Howto configure asterisk with AVM ISDN card
Claudio Condolf wrote: Hi guys, I'trying to setup Asterisk with an AVM Fritz PCI ISDN card with no success. I have an IP phone from Micronet and I want to receive and place calls over ISDN. But I'm not sure if I need the binary CAPI from AVM or isdn4linux. I've tryed with redhat 9 and with Suse 8.2. Cold someone help me? Thank you very much. I did the following on RH9.. First goto ftp://ftp.avm.de/cardware/fritzcrd.pci/linux/suse.82/ and read the info.txt and download the tar.gz file.. Follow the instructions to compile and install the module.. Then goto http://www.junghanns.net/asterisk and download chan_capi Follow the instructions to compile and install.. This should get you to a point where you can load the capi driver for the ISDN card (capi init from a command line) and load the capi channel driver when Asterisk loads.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX experiment - Should this have worked better?
Should this have worked better?
1) I register my * server with iaxtel.com as user2:[EMAIL PROTECTED]
2) I call to my * server using X-Lite from a Win2k box and dail
the 1700 iax number associated with user2 above
3) my * server corectly routes the call thru
_91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
4) sure enough I here the main menu from my own * server.
It seems to have worked but...
The throuble is the __very poor__ sound quality. It has long gaps of
missing speech, some jittery speech and then it disconnects. I know
the
sound is making a round trip to iaxtel.com andback and then oover
a local SIP link back to X-Lite but but the result should, I think be
understandable.
I am also sending 800 numbers to iaxtel,com. with this:
_91700NXX,1,Dial(IAX2/${IAX...
and the sound quality is not that bad, not perfect but but fair
and certanly without dropouts or gaps.
So what's up does two hops over IAX kill the audio? Should I
messaround more with jitter buffer settings?
I'm on a DSL line, in back of a Linsys router/firewal
My 2.6Ghz P4 nerver shows more than 1% CPU utilization
These calls take
=
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
KG6OMK
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Re: [Asterisk-Users] IAX experiment - Should this have worked better?
- Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 10:13 AM Subject: [Asterisk-Users] IAX experiment - Should this have worked better? Should this have worked better? 1) I register my * server with iaxtel.com as user2:[EMAIL PROTECTED] 2) I call to my * server using X-Lite from a Win2k box and dail the 1700 iax number associated with user2 above What Codec are you using between X-Lite and * As be sure to turn of ILBC and Speex as they are not properly working between X-Lite *. GSM or 711 should be fine. Greetings, Tjardick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P setup in Switzerland
I have some news ... after a bit of tweaking, the following seems to work with Swisscom : span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 I still have a problem : Incoming calls from PRI work well, but outgoing call don't : I dial from a Grandstream 101 -- Executing Dial(SIP/22-9129, Zap/g1/234234234) in new stack -- Called g1/234234234 -- Zap/1-1 is ringing Well ... I hear a ringing tone in the Grandstream, but the called number doesn't ring at all ... What can I do ?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supported dialogic hardware ?
Hi, Anyone knows if the Dialogic hardware D4PCIEURO (http://www.intel.com/network/csp/products/5795web.htm) is supported by Asterisk. Regards, Jorge [EMAIL PROTECTED]
Re: [Asterisk-Users] my phone shows asterisk
Hi! My setup is: pstn X100PASTERISKSNOM 200 thanks miklos - Original Message - From: Gerry Boudreaux [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:12 PM Subject: Re: [Asterisk-Users] my phone shows asterisk What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos ___ 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] my phone shows asterisk
listas iPfone wrote: Hi! My setup is: pstn X100PASTERISKSNOM 200 thanks miklos - Original Message - From: Gerry Boudreaux [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:12 PM Subject: Re: [Asterisk-Users] my phone shows asterisk What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos I am guessing your don't have CallerID enabled from your telco.. If you do it is probably incompatible.. If you really hate the asterisk showing on your screen then use the SetCallerID and SedCIDName commands on your inbound calls.. eg.. exten = s,1,SetCallerID(555 4321) exten = s,2,SetCIDName(Inbound Call) exten = This will obviously be statically set and will not show the CallerID of the person that is calling but it wil get rid of the word asterisk on your screen.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Asterisk-User] Howto get the Caller Phonenumber ?
Hello, Can anyone suggest us how to got the phonenumber of the caller. In the environment variable, I just see the ip of the gateway. Environment: 'agi_callerid' is 'XXX.xxx.XXX.xx' Should I do some changes in the GW conf, or is it just not possible when we got a call from PSTN ?!? Thx in advance, Ares ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one-way audio
Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person who is being called. This appears to be intermittent, with occasional times where conversations can be held, but what sounds like silence suppression on the side of the caller. They hear audible clicks and then silence, and assume the person they have called has hung up. On the side of the person being called there are no audible problems at all (except for the fact that most of the time there is nothing being heard on their side). so the link between the two machines is using a 512k connection at one end, and a 2meg connection at the other. an IAX2 trunk between them. the only thing i can think of as a potential cause of this problem is that one is using a cvs thats more recent than the other. im worried about updating the older machine without being sure that its not going to introduce problems as there are many other applications and services running on that server. any help would be much appreciated duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] modem connection over handy?
Hi all, does anybody know if it is possible to make a modem connection (voice) through * over a handy which is connected to the RS232 port ? I get following messages when * starting: WARNING[16384]: File chan_modem.c, Line 356 (modem_setup): Modem reset failed: (No Response) WARNING[16384]: File chan_modem.c, Line 735 (mkif): Unable to configure modem '/dev/ttyS0' ERROR[16384]: File chan_modem.c, Line 871 (load_module): Unable to register channel '/dev/ttyS0' WARNING[16384]: File loader.c, Line 301 (ast_load_resource): chan_modem.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 347 (load_modules): Loading module chan_modem.so failed! Thanks for help, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my phone shows asterisk
Hi! Thanks for the advice i will do it. There is a way to know if the CallerID enabled from my telco is compatible with asterisk? regards Miklos - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 8:08 AM Subject: Re: [Asterisk-Users] my phone shows asterisk listas iPfone wrote: Hi! My setup is: pstn X100PASTERISKSNOM 200 thanks miklos - Original Message - From: Gerry Boudreaux [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:12 PM Subject: Re: [Asterisk-Users] my phone shows asterisk What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos I am guessing your don't have CallerID enabled from your telco.. If you do it is probably incompatible.. If you really hate the asterisk showing on your screen then use the SetCallerID and SedCIDName commands on your inbound calls.. eg.. exten = s,1,SetCallerID(555 4321) exten = s,2,SetCIDName(Inbound Call) exten = This will obviously be statically set and will not show the CallerID of the person that is calling but it wil get rid of the word asterisk on your screen.. Later.. ___ 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] Problems with Devkit Lite setup
At one point I did have this kit working but since upgrading to the latest Asterisk, it no longer seems to. I had the following problems after several reinstallations: - USB adaptor had a proper dialtone, asterisk recognised the pickup, but pressing keys on the handset had no effect - USB adaptor produced a strange horrible tone, not a dialtone; pressing keys has no effect - USB adaptor silent; pickup not recognised by Asterisk So you can see my situtation has gradually deteriorated with each tinkering/CVS-reinstall of the system. What I don't understand is what Linux or other elements do I need to get the USB adaptor to work? I am running RedHat 9 with the 2.4.20-20.9 kernel. Apart from these problems, I also noticed the following when using the USB adaptor: - unplugging and plugging the handset into the line side of the adaptor killed the adaptor; have to restart the machine to clear the problem. - unplugging the USB adaptor from the USB port causes all manner of problems. 'rmmod'-ing and 'insmod'-ing astrisk related modules doesn't help. Running asterisk or 'ztcfg' always complains of problems on the USB adaptor channel. Restart of machine needed. - sometimes problems with the computer's audio system ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one-way audio
Im experiencing a problem with a current setup and I've run out of ways to debug it and come to a resolution. I have two E100P's in a machine which is routing traffic over the internet to a machine that has 1 E400P connected to the PSTN. Clients are able to make calls successfully but when the call is connected experience one-way audio. They cannot hear anything said by the person who is being called. This appears to be intermittent, with occasional times where conversations can be held, but what sounds like silence suppression on the side of the caller. They hear audible clicks and then silence, and assume the person they have called has hung up. On the side of the person being called there are no audible problems at all (except for the fact that most of the time there is nothing being heard on their side). so the link between the two machines is using a 512k connection at one end, and a 2meg connection at the other. an IAX2 trunk between them. the only thing i can think of as a potential cause of this problem is that one is using a cvs thats more recent than the other. im worried about updating the older machine without being sure that its not going to introduce problems as there are many other applications and services running on that server. ooh replying to my own posts again. bad duncan. some more information on the problems being encountered. if i have an IAX2 Trunk - the person making the call cant be heard but the person recieving the call can hear them if i have a standard call over IAX2 the person recieving the call cant hear anything but can be heard by the person making the call if i have a standard IAX call - both sides can hear each other fine. any ideas where the problem lies based on this new information? duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my phone shows asterisk
listas iPfone wrote: Hi! Thanks for the advice i will do it. There is a way to know if the CallerID enabled from my telco is compatible with asterisk? regards Miklos I guess if it is enabled and it does not work then chances are that it is not going to be compatible.. Post who your Telco is and maybe there is someone else who knows.. later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one-way audio
person recieving the call can hear them if i have a standard call over IAX2 the person recieving the call cant hear anything but can be heard by the person making the call if i have a standard IAX call - both sides can hear each other fine. any ideas where the problem lies based on this new information? You should be able to see useful information on the console. One-way audio with IAX2 is, generally speaking, not possible -- especially not without trunking. Turn off the jitter buffer in your iax.conf to be on the safe side. Also, tcpdump is your friend. Remember, IAX2 only uses a single port, so you should be able to see traffic in both directions. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller Id AGI Script
As you my be aware the X100p cannot collect uk caller id, now I have a modem and a perl script that creates a file /etc/asterisk/callerid.txt on each incoming call in the format number,date,time,name over writing each time a new call comes in I can asterisk read this file and populate the callerid for internal phones and cdr? I think it can be done with AGI but don't know where to start could someone point me in the right direction Thanks in advance for your help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Banks
Mark Evans wrote: Hi All Can you's give me your thoughts on the best channel banks to use? Which are the easist to setup and which are the most reliable. Thanks Mark You may know already but the vast majority of channel banks are T1 only and typically only available in the US.. At least this is what I found when I was looking at using one.. Of course the dual mode card from Digium removes alot of the problem now in that you can have one or two ports of T1 to a CB and another port E1 to the PSTN.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Banks
-- Original Message -- From: Mark Evans [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 10 Oct 2003 15:19:36 +0200 Hi All Can you's give me your thoughts on the best channel banks to use? I have use Adtran 750 and Adtran 600's. They are very easy to work with and seem to work just out of the box! There not cheap but work every time! Which are the easist to setup and which are the most reliable. Thanks 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
RE: [Asterisk-Users] Channel Banks
Well I disagree, there are numerous companies providing E1 channel banks, my personal favourite is J-tech of which I can find the damn link to their page for now ... Digging ... A quick google with e1 channel banks also found: http://www.valiantcom.com/vcl_cb/vcl_cb.html -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 10 October 2003 15:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Banks Mark Evans wrote: Hi All Can you's give me your thoughts on the best channel banks to use? Which are the easist to setup and which are the most reliable. Thanks Mark You may know already but the vast majority of channel banks are T1 only and typically only available in the US.. At least this is what I found when I was looking at using one.. Of course the dual mode card from Digium removes alot of the problem now in that you can have one or two ports of T1 to a CB and another port E1 to the PSTN.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller Id AGI Script
Robert, As you've already got the modem recording the callerid etc, I shall presume you're perl and bash knowledge should be up to scratch. All you gotta do is, using AGI and perl, run a perl script which parses the info from the file and simply use AGI - setcallerid to pass the number into asterisk. Good intro to AGI here http://home.cogeco.ca/~camstuff/agi.html AGI/perl related stuff here http://asterisk.gnuinter.net/ If you need some further help getting your script working, mail me offlist. Now, I've got some questions for you :) What modem model are you using and would you like to share your script which captures the uk callerid, as this has been a major shortcoming for asterisk with UK users. HTH Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
No, you actually don't need to use a context in the peer. Asterisk will ? leave it up to the far end to decide what context to use. We use it to avoid any possibility of confusion in the process, but it is not necessary. In fact, I just verified this with the master himself and we will no longer tell our customers to use a context in their peer. And, given that a context= entry is not used for a peer, only for a user, it also goes to say that if you use a friend, context is still only used for the *inbound* portion of it. So I just don't really see why you so strongly recommend against friend entries as opposed to having both a user and a peer entry that are otherwise the same... Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with DTMF 'looping' / mis-dials (X100P card)
In the sample zaptel.conf file (see the asterisk source directory) there is the following: ; ; If you are having trouble with DTMF detection, you can relax the ; DTMF detection parameters. Relaxing them may make the DTMF detector ; more likely to have talkoff where DTMF is detected when it ; shouldn't be. ; I'm not going to post the actual parameter since you can look it up in the sample config file that comes with Asterisk easy enough. Take the time to read thru the file just in case there are some other paramters that you might find useful. Just don't use the callprogress or busydetect options. On Fri, 2003-10-10 at 00:27, Sam S wrote: Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything 300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct extension number, delete a voicemail message, etc. Any ideas why this is happening, or how to fix it? I searched the mailing list back to 7/1/03 but found no mention. Here's my * setup. Note presence of Vodavi Starplus DHS phone system in call path. Pentium II -350 / Redhat 9.0 / (3) X100P cards X100P cards are connected to an Analog SLT adaptor, which goes to a digital ports on a Vodavi StarPlus DHS phone system. (So call path is: PSTN -- Vodavi Starplus -- Analog SLT adaptor -- X100P card.) Asterisk CVS-08/29/03-09:23:49 I have not updated to the most recent CVS because of various problems I've seen cropping up on the bug tracking site.. Something tells me this is not CVS-related but perhaps something to do with the Vodavi. Any suggestions? DTMF parameters I can tweak? Here's an example of what Voicemail2 does when I hold down the 7 key while listening to a message: (Flip-flop period is about 3-4 cycles per second.) -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller Id AGI Script
Hi Dave I've not completed the script yet, But you may not like this but I've had to use a win98 box for the zoom 3025C (important its the C model), the zoom modem is the only internal one I've found that can do uk caller id (but its not supported on the linux driver), and is still available. ( I bought it from PC world for £25) and I'm going to hack this script http://www.anderbergfamily.net/ant/caller id/ to produce the file required There are linux drivers for the card (www.linuxant.com) but the UK caller ID is not implemented (yet) but if lots of us post a reply to the message board and maybe they'll implement, I've already sent Linuxant the requested info http://www.linuxant.com/pipermail/hcflinux/2003q3/001056.html Robb Quoting Dave Wilson [EMAIL PROTECTED]: Robert, As you've already got the modem recording the callerid etc, I shall presume you're perl and bash knowledge should be up to scratch. All you gotta do is, using AGI and perl, run a perl script which parses the info from the file and simply use AGI - setcallerid to pass the number into asterisk. Good intro to AGI here http://home.cogeco.ca/~camstuff/agi.html AGI/perl related stuff here http://asterisk.gnuinter.net/ If you need some further help getting your script working, mail me offlist. Now, I've got some questions for you :) What modem model are you using and would you like to share your script which captures the uk callerid, as this has been a major shortcoming for asterisk with UK users. HTH Dave ___ 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] Cisco 7914
I've never been able to try it for myself as the few 7914s I have laying around here have no interface cable and I am unable to find the pinout. Even TAC couldn't help me :( I'll find out the pinout and post it tomorrow :) As a side note, the skinny image does support them, will it work with chan_skinny? I just looked at the cable, it's a simple straight trough, 6pin rj11 cable. You do need a power supply brick for the first expansion module, the second one draws it's power through the serial cable from the first one. -- Yifang Dai | eFax: (847)628-0255 |Debian GNU/Linux [EMAIL PROTECTED] | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] telefonica sp brazil caller id problem
Hi I have problems with caller id in my line from telefonica sp brazil, anyone knows if there is any problem with this telco caller id and asterisk? thanks miklos - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 10:10 AM Subject: Re: [Asterisk-Users] my phone shows asterisk listas iPfone wrote: Hi! Thanks for the advice i will do it. There is a way to know if the CallerID enabled from my telco is compatible with asterisk? regards Miklos I guess if it is enabled and it does not work then chances are that it is not going to be compatible.. Post who your Telco is and maybe there is someone else who knows.. later.. ___ 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] 7940/60 TFTP Problem
OK, I got it figured out. Here is the process to fix this problem in the future. The key here on MGCP phones was to recognize that they only understand the 8.3 naming format until a few revs up. What seemed like corruption was the Cisco phones inability to understand bigger than 8.3 names, such as what Cisco uses for the SIP images. Here is the step-by-step: 1. Upgrade to a newer version of the MGCP code. I went to: P0M3-03-0-00.bin However this does not work out of the box,I had to copy the file to P0M30300.bin to adhere to the 8.3 name standard. Modify the OS79XX.TXT to say only the following: P0M30300 This process apparently reorganizes the image memory mapping on the phone to make room for the larger image to come. 2. Once the upgrade is complete upgrade to a newer MGCP code. I went to : P0M3-04-4-00.bin I again copied the file to P0M30440.bin and modified OS79XX.TXT to: P0M30440 3. Only now am I ready to upgrade to the SIP 5.3 image. I'm not sure if I still had to adhere to the 8.3 naming convention, I did it just to be safe, but I am now running on 5.3 SIP. SOme screwy things happened with my config. I got garbage on my domain name field that I could not back space to erase. I just reset to factory to adress this issue. Thanks for everyone's help and I hope this document is useful to someone in the future. Babak Rich Adamson wrote: Babak, I don't have a clue where these came from. ;) Rich Would someone have a 2.2 SIP that I could try to keep in accordance to Rich's methodology? Babak -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream wallmount??
Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Config
Title: Leterhead What do you have configured in your /etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci this should show adevice Tiger Jet Network Inc. if the pci bus recognized the card. - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:05 PM Subject: [Asterisk-Users] X100P Config Hiya all, I have just received my X100P telco card and I dont seem to be able to talk to it. I am a bit of a numpty on Linux being from the Windows (wash my mouth with soap and water) background, so any help would be appreciated. I have checked under YaST2 and I think it can see the card, but not sure. My * box is talking between 2 Grandstream phones no probs but now I would like to talk to the outside world. Thanks in anticipation. Dave
Re: [Asterisk-Users] Grandstream wallmount??
Dave Weis wrote: Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. No, you aren't the only one. The absence of the hook has been the source of considerable amusement in our lab! Stephen Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEWBIE looking for advice.
We have a few boxes out there with T100P's and Adtran 750's configured similar to what you suggest, and use adsi phone for extensions. It has been almost trouble free. We have also gone back and mixed in SIP snom, grandstream, and cisco phones with reasonable success. We have not had an echo problem with this setup, and the programmabillity of the adsi phones let us add simple features such as internal directories, and speed dials to commom asterisk features such as the external paging extension. Its what I use at work also. Callerid works fine, as does faxing, transferring, and 3way calling. The only feature we miss from our old system is station to station paging, but a cheap stereo and speakers gets us by... Chad Sawyer - Original Message - From: Joshua Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:39 PM Subject: [Asterisk-Users] NEWBIE looking for advice. All, Thanks for taking time to read. I am wanting someone to tell me that I have a clue here and that the following proposed setup would (possibly) work. Currently existing are 4 pots lines with qty. 10 ATT 954/854 phones. Would like to convert to asterisk and at the same time do a mild feature upgrade. Proposing to get: PIII 800MHz or better =256MB RAM Linux box running Asterisk Adtran TA750 w/ 6 FXS Boards (24 channels w/ T1) Get a Quad FXO board and swap it with one of the FXS boards. Put a T100P or TE410P(expandability) in the Linux box and connect with crossover T1 Cable to the Adtran. Run the current 4 POTS lines into the FXO's on the TA750. Wire the office phones to the FXS ports (now 20 of them since board swap) Replace ATT *54 phones with POTS phones, hopefully with a flash button. Configure Asterisk, and enjoy !! Would this be: 1) A sane solution? 2) Reliable/Hassle Free--compared to SIP phones, H.323, IAX etc.? 3) Could I expect fax/data/callerID to function? 4) Are there problems in tweaking the flash configuration, to get things like hold, transfer, 3 way etc. to work. Thanks in advance for any/all advice. Joshua ___ 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] Grandstream wallmount??
On Fri, 2003-10-10 at 20:49, Stephen R. Besch wrote: Dave Weis wrote: Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. No, you aren't the only one. The absence of the hook has been the source of considerable amusement in our lab! No, but I've used the notches to mount mine on a 30° slope, makes it mush easier to see the display. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream wallmount??
hehe... I've been thinking of some manual modifications. quote who=Dave Weis Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - H323 GAteway
Mireia Munoz de jesus wrote: Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register a SIP number in a H.323 gatekeeper? I know that with NetMeeting I can make calls peer-to-peer dialing an IP... but if the softtelephone of the other terminal is an SIP UA that is not going to work, is it? Please any help will be welcome. Regards, Mireia What gatekeeper do you use. It seems that is programed to make outgoing calls only to registered h.323 users. Just program it to forward unknown number to the asteris (or switch everything to SIP) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream wallmount??
Dave Weis wrote: Am I the only one that has noticed there is no way to wallmount a Grandstream phone? There are screw notches on the back, but no hook to hold the handset in. knife, rasp, glue(strong), wrecks from plastic lighter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ISA tormenta card message]
I am getting the following messages that seem to be coming from Asterisk. In the system there are no ZAPTEL cards installed. I did uncomment ztdummy in the Makefile in /usr/src/zaptel before running make install. Any ideas on how to get rid of this message. I looked through all the config files (installed the sample ones then modified sip.conf, extensions.conf and voicemail.conf, rest are as installed) but did not find anything that looked right. Can someone please point me toward what I am overlooking? Thanks Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple SIP users on one phone?
Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's about to pick up the phone who the call is for. Maybe change the caller ID field or something? In an ideal world, the hardware would have two lines, one with caller ID, one with the name of the person they're trying to call. Additionally, a person should be able to field a call, decide they can't deal with it, and push a button that redirects the caller to the appropriate person's voicemail (based on the incoming DNID). Is any of this possible? Phone cost per handset should be as low as possible, as per usual. :) I have no experience of SIP hardphones, so don't know how much/what information about the call they're capable of displaying. Regards, -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Config
Title: Leterhead Hi, I can see the card with a cat /proc/pci. I dont seem to have a zaptel.conf file in the etc directory. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Glenn Dalgliesh Sent: 10 October 2003 19:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config What do you have configured in your /etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci this should show adevice Tiger Jet Network Inc. if the pci bus recognized the card. - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:05 PM Subject: [Asterisk-Users] X100P Config Hiya all, I have just received my X100P telco card and I dont seem to be able to talk to it. I am a bit of a numpty on Linux being from the Windows (wash my mouth with soap and water) background, so any help would be appreciated. I have checked under YaST2 and I think it can see the card, but not sure. My * box is talking between 2 Grandstream phones no probs but now I would like to talk to the outside world. Thanks in anticipation. Dave Registered Office: - 23 First Street, Low Moor, Bradford, West Yorkshire, BD12 0JQ. Company Registration Number: - 03807643. VAT Registration Number: - 734-3363-42 Telephone / Fax: - 44 (0) 7092 154039. SIP_Phone: - 1 (747)669 1957 http://www.codepipe.ltd.uk / http://www.codepipe.com / E-Mail: - [EMAIL PROTECTED]
[Asterisk-Users] ISDN PRI CallerID NAME question
Is Caller ID Name actually ever set by asterisk. I am looking at the PRI debug messages and do not see it going out. I doo see the number being sent out. Excuse me if I am a little off here but am I supposed to be able to set the Text on a PRI Span? -Nicholas Romero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Config
LeterheadThe FAQ at digium explains how to do it: http://www.digium.com/index.php?menu=faq#Configuration_7 - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 5:29 PM Subject: RE: [Asterisk-Users] X100P Config Hi, I can see the card with a cat /proc/pci. I don't seem to have a zaptel.conf file in the etc directory. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Glenn Dalgliesh Sent: 10 October 2003 19:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config What do you have configured in your /etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci this should show a device Tiger Jet Network Inc. if the pci bus recognized the card. - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:05 PM Subject: [Asterisk-Users] X100P Config Hiya all, I have just received my X100P telco card and I don't seem to be able to talk to it. I am a bit of a numpty on Linux being from the Windows (wash my mouth with soap and water) background, so any help would be appreciated. I have checked under YaST2 and I think it can see the card, but not sure. My * box is talking between 2 Grandstream phones no probs but now I would like to talk to the outside world. Thanks in anticipation. Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ** Adtran 750, 5xQuad FXS, 1xQuad FXO, Pentium 500MHz, 10GB hard Drive, 384MB RAM..Will that do?
Hello Everyone, I have been thinking about starting Asterisk efforts for quite some time. Finally I got myself into it. Just received a T100P from Digium and I had already secured an Adtran 750TA with FXS and FXO cards. My Goal: Create a PBX based on Asterisk and Digium hardware with following configuration: 4 FXO (now) 20 FXS (now) IAX connectivity to other Asterisks outside (Future) VoIP explorations (Future) My hardware configuration: Adtran 750 TA with 1-Quad FXO and 5-Quad FXS. Digium T100P. Pentium 500 MHz 10 GB hard-drive. 384MB RAM. 2-Network interface cards. CD-ROM/FLOPPY. Total of 4 PCI slots.(2 available) I suppose this is enough to get me a world class PBX ? Any thoughts? I have been reading a lot of posts and messages and this seems to be very popular configuration amongst the Asterisk users. So to all of you out there, who have gone through similar set-up, please help me get started. There is a lot that can be done with Asterisk I am willing to explore all the features but I need basic help to get started and to boost my confidence in Asterisk. Assuming minimal familiarity with Linux and almost no familiarity with Asterisk, is there a step-by-step guide for the discussed configurations. If there is no documentation already, i would love to create one once I am all installed and ready. Please share any supporting literature (such as a set of .config files) for the purpose with me. You can email me directly at [EMAIL PROTECTED] (yes that is a real address but fools the spammers, I hope :-) Thanks a lot in advance and I look forward to hearing from all of you. Once I am all set-up, as promised earlier, I will create a step-by-step (starting form linux installation) guide for future starters. If there is any such guide already out there, please share it with me. Ricky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P Config
Thanks for that. I will check there in the morning. Chow for now. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of TeleSIP Sent: 10 October 2003 23:38 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config LeterheadThe FAQ at digium explains how to do it: http://www.digium.com/index.php?menu=faq#Configuration_7 - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 5:29 PM Subject: RE: [Asterisk-Users] X100P Config Hi, I can see the card with a cat /proc/pci. I don't seem to have a zaptel.conf file in the etc directory. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Glenn Dalgliesh Sent: 10 October 2003 19:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Config What do you have configured in your /etc/zaptel.conf * /etc/asterisk/zapata.conf file. Also, submit a cat /proc/pci this should show a device Tiger Jet Network Inc. if the pci bus recognized the card. - Original Message - From: David J Carter To: [EMAIL PROTECTED] Sent: Friday, October 10, 2003 2:05 PM Subject: [Asterisk-Users] X100P Config Hiya all, I have just received my X100P telco card and I don't seem to be able to talk to it. I am a bit of a numpty on Linux being from the Windows (wash my mouth with soap and water) background, so any help would be appreciated. I have checked under YaST2 and I think it can see the card, but not sure. My * box is talking between 2 Grandstream phones no probs but now I would like to talk to the outside world. Thanks in anticipation. Dave ___ 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] IAX Not working between machines
I setup two machines to talk to each other with IAX and it does not seem to work. When a call comes into one machine and transfers it to the other, the machine that is transferring to the other one shows: -- Accepting AUTHENTICATED call from 65.127.126.42, requested format = 4, actual format = 4 -- Executing Dial([EMAIL PROTECTED]/4, IAX2/USERNAME:[EMAIL PROTECTED]/320|90|r) in new stack -- Called USERNAME:[EMAIL PROTECTED]/320 WARNING[9226]: File chan_iax2.c, Line 4152 (socket_read): Call rejected by 66.156.244.159: No authority found And the machine that should receive the call says: *CLI NOTICE[8201]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect attempt from 69.41.227.70 Any clues? I checked the contexts in IAX.conf and they should be correct. If anything I would expect a not found sort of message.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P check for Dialtone
Is there any way to make the X100P to check for a dialtone before dialing? My roommate is tired of hearing random dialing in his ear when he's on the phone (same circuit as the X100P). I know one solution is to just put in a TDM400P for the house, but I'd rather get away w/o having to do that. Thanks, Jared ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Not working between machines
Andrew Joakimsen wrote: I setup two machines to talk to each other with IAX and it does not seem to work. When a call comes into one machine and transfers it to the other, the machine that is transferring to the other one shows: -- Accepting AUTHENTICATED call from 65.127.126.42, requested format = 4, actual format = 4 -- Executing Dial([EMAIL PROTECTED]/4, IAX2/USERNAME:[EMAIL PROTECTED]/320|90|r) in new stack -- Called USERNAME:[EMAIL PROTECTED]/320 WARNING[9226]: File chan_iax2.c, Line 4152 (socket_read): Call rejected by 66.156.244.159: No authority found And the machine that should receive the call says: *CLI NOTICE[8201]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect attempt from 69.41.227.70 Both of these errors mean the same exact thing, the calling party does not have the authority to call the called party. You need something like this: [USERNAME] type=user secret=SECRET context=the_appropriate context_here_containing_the_320_exten on the 69.41.227.70 machine. Also, if your not careful, calling like that would cause a Native Bridge to occur, this may or may not be something you want, depending on if you wish to bill for this telephone call or not. Thank you, drive-thru, Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Setup
Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron. - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Not working between machines
I double checked the contexts and they are all using the same context right now. I also just notied this message: NOTICE[9226]: File chan_iax2.c, Line 2855 (register_verify): Peer USERNAME is not dynamic (from 66.156.244.159) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Friday, October 10, 2003 8:25 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX Not working between machines Andrew Joakimsen wrote: I setup two machines to talk to each other with IAX and it does not seem to work. When a call comes into one machine and transfers it to the other, the machine that is transferring to the other one shows: -- Accepting AUTHENTICATED call from 65.127.126.42, requested format = 4, actual format = 4 -- Executing Dial([EMAIL PROTECTED]/4, IAX2/USERNAME:[EMAIL PROTECTED]/320|90|r) in new stack -- Called USERNAME:[EMAIL PROTECTED]/320 WARNING[9226]: File chan_iax2.c, Line 4152 (socket_read): Call rejected by 66.156.244.159: No authority found And the machine that should receive the call says: *CLI NOTICE[8201]: File chan_iax2.c, Line 4060 (socket_read): Rejected connect attempt from 69.41.227.70 Both of these errors mean the same exact thing, the calling party does not have the authority to call the called party. You need something like this: [USERNAME] type=user secret=SECRET context=the_appropriate context_here_containing_the_320_exten on the 69.41.227.70 machine. Also, if your not careful, calling like that would cause a Native Bridge to occur, this may or may not be something you want, depending on if you wish to bill for this telephone call or not. Thank you, drive-thru, Jeremy McNamara ___ 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] multiple SIP users on one phone?
Interesting problem: An organization has departments. Each department has a single phone. Each department has multiple people. Each person within the organization has a direct dial incoming number. It's easy to set * up so that multiple DDIs get mapped to the same extension. What I'm wondering is if there's any way, with reasonably priced hardware, to notify the person who's about to pick up the phone who the call is for. Maybe change the caller ID field or something? In an ideal world, the hardware would have two lines, one with caller ID, one with the name of the person they're trying to call. Additionally, a person should be able to field a call, decide they can't deal with it, and push a button that redirects the caller to the appropriate person's voicemail (based on the incoming DNID). Is any of this possible? Phone cost per handset should be as low as possible, as per usual. :) I have no experience of SIP hardphones, so don't know how much/what information about the call they're capable of displaying. Regards, -- Alastair Maw The Cisco 7960 supports six lines, meaning six DID's, and inbound calls will flash the appropriate line icon on the LCD screen. Similarly, if the correct line appearance button is pressed before dialing, the outbound SIP call will appear to originate from that particular line. Each line also has it's own MWI indicator on the screen in the form of a flashing envelope. Price is ~$300 these days for used 7960 phones, and then you'd need to contact Cisco if the phones did not already have the SIP image on them. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P Phones Configuration
Below you will find, what I believe to be a typical setup with a T100P
card. My question is -
1. Is this correct?
2. What kind of phones would be needed here... (Would you have to use
Digital phones) And if so what would you recommend.
PRI/T1-
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| Channel Bank |
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| |Amphenol | 24 Port Patch |
| --| Panel|
|||
-||
||
| * Server | || ||
| T100P | || ||
|| || ||
- Phone Phone Phone Phone
Thank you,
Geoff
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Re: [Asterisk-Users] IAX Not working between machines
try adding: host=dynamic to the definition of the user in iax.conf Andrew Joakimsen wrote: I double checked the contexts and they are all using the same context right now. I also just notied this message: NOTICE[9226]: File chan_iax2.c, Line 2855 (register_verify): Peer USERNAME is not dynamic (from 66.156.244.159) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Marketing Digium/Asterisk
The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies who would benefit from IP trunking - perhaps by industry segment. ie what type of company is likely to have multiiple locations with lots of interoffice communications. Also, I would love to target Centrex customers. I put in an Asterisk system for a local company that had been on Centrex. In short, it will save them $18,000.00 per year (and did pretty well for myself too:)). Any way to ID centrex customers? I'd be glad to share any tips ideas etc. as well if there is any interest. Just a thought. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Phones Configuration
Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? Possibly. Depends on if you use a channel bank that can do add/drop and you're not using a PRI. You'll take your incoming T1 and go into 1 T100P and use another T100P to feed out to your channel bank...or you can get a T400P and just have one card in the system. 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. You can use anything from a $9 WalMart phone to a $300 ADSI analog phone. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Marketing Digium/Asterisk
You need to remember you are looking at it different to what they would. Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Sent: Saturday, 11 October 2003 1:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Marketing Digium/Asterisk The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies who would benefit from IP trunking - perhaps by industry segment. ie what type of company is likely to have multiiple locations with lots of interoffice communications. Also, I would love to target Centrex customers. I put in an Asterisk system for a local company that had been on Centrex. In short, it will save them $18,000.00 per year (and did pretty well for myself too:)). Any way to ID centrex customers? I'd be glad to share any tips ideas etc. as well if there is any interest. Just a thought. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ 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] Marketing Digium/Asterisk
comments inline -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, October 10, 2003 11:05 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk You need to remember you are looking at it different to what they would. Regards Mick Yes, this is true, however, I didn't mean to imply that I wouldn't aggressively market the benefits to the potential customer, but rather that I wanted to find a way to id companies who have the most to benefit from this type of solution. Thanks for the reminder though,I need it as I am not much of a salesman. Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Sent: Saturday, 11 October 2003 1:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Marketing Digium/Asterisk The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies who would benefit from IP trunking - perhaps by industry segment. ie what type of company is likely to have multiiple locations with lots of interoffice communications. Also, I would love to target Centrex customers. I put in an Asterisk system for a local company that had been on Centrex. In short, it will save them $18,000.00 per year (and did pretty well for myself too:)). Any way to ID centrex customers? I'd be glad to share any tips ideas etc. as well if there is any interest. Just a thought. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ 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] Marketing Digium/Asterisk
That's OK Easy to get involved and lose sight of the reason these customers A couple of companies that I found to approach were Companies with multiple offices With lots of calls between them and spending thousands of dollars a month. Just food for thought Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Sent: Saturday, 11 October 2003 1:48 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk comments inline -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, October 10, 2003 11:05 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk You need to remember you are looking at it different to what they would. Regards Mick Yes, this is true, however, I didn't mean to imply that I wouldn't aggressively market the benefits to the potential customer, but rather that I wanted to find a way to id companies who have the most to benefit from this type of solution. Thanks for the reminder though,I need it as I am not much of a salesman. Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hester Sent: Saturday, 11 October 2003 1:31 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Marketing Digium/Asterisk The benefits of * are obvious so that part of the marketing an * solution is easy. Anybody care to share ideas on how to target companies who would benefit most from */Digium? It seems to me that it would be an easy sell to small/medium companies who need advanced features such as ip trunking, IVR, Conference bridging, etc., etc. I would like to find a way to identify multi-location companies who would benefit from IP trunking - perhaps by industry segment. ie what type of company is likely to have multiiple locations with lots of interoffice communications. Also, I would love to target Centrex customers. I put in an Asterisk system for a local company that had been on Centrex. In short, it will save them $18,000.00 per year (and did pretty well for myself too:)). Any way to ID centrex customers? I'd be glad to share any tips ideas etc. as well if there is any interest. Just a thought. Sincerely, Andy Hester Consero (817)375-1244 (817)937-7977 ___ 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] Marketing Digium/Asterisk
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Friday, October 10, 2003 11:28 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Marketing Digium/Asterisk That's OK Easy to get involved and lose sight of the reason these customers A couple of companies that I found to approach were Companies with multiple offices With lots of calls between them and spending thousands of dollars a month. Just food for thought Regards Mick Yes, but what type of business did they do? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Setup
My config that works for number 1 is below. Everything works including the voice mail waiting light. All of this for * was copied from or based on: http://www.automated.it/guidetoasterisk.htm. This is an EXCELLENT getting started site. Can't help you with #2 but am sure others can. sip.conf for extension 2000 [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=9overthruster7 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it extensions.conf exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Voicemail(u2000) Budge Tone config: SIP Server: 192.168.0.110 (my * box) SIP Userid: 2000 (userid is same as extension Authenticate ID: 2000 Authenticate password: 9overthruster7 Send DTMF: Via SIP info (in order for the dtmf to be recognized by voicemail) Hi People, Ok i've tried everything I can think of but cant get this to work. Can someone please give me an example of their sip.conf settings and also the details of the settings in their grandstream phone to allow: 1. Grandstream phone to register with asterisk when on same lan. 2. Grandstream phone to register with asterisk when phone is behind a nat. Regards, Aaron. - This mail sent through IMP: http://horde.org/imp/ ___ 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
