Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
Hi Andrew, Very very very STRANGE! It seems that I have not received this mail from Michael, even it is posted to the distribution list and this is not the only one?!?! Someone else with this problem? Well, I saw at least 4 messages recently from the asterisk-users mailing list, that spamassassin scored over 5. I believe the sender's SMTP relay was in the various open relay databases. So you may have filtered some messages if you're using a spam filter and haven't whitelisted lists.digium.com. I have no anti-spam app installed on my PC nor at the provider. Another strange thing (only) with the mails from Michael: I have my main PC and my notebook using Outlook Express, with the option to keep the mails on the server. If I read the Michael's mail on one of them, then it will never be available on the other one (as it was deleted from the server) And all this... ONLY with the Michael's mails received through this distribution list... BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question about incoming/outgoing
Larry Black wrote: [hardware] type=friend callerid=Hardware Phone 5 secret=phone echocancel=yes host=dynamic dtmfmode=rfc2833 context=sip My standard config for GS phones on the same LAN as the Asterisk server is.. [hardware] type = friend callerid = Hardware Phone 5 secret = phone host = dynamic dtmfmode = info context = sip To specify codecs you could add.. disallow = all allow = ulaw allow = alaw I can't see from you phone config why you are having this problem.. I know GS are close to a major upgrade on the firmware so maybe this will help.. Also a suggestion, you may find it easier to manage if you name the phones by their extension number rather than a name.. it will make your extensions.conf a little easier to create and modify.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar
Sent: Tuesday, November 18, 2003 10:12 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Updated iaxComm binaries available for
WinXP,
Red Hat 9.0
On Tue, 18 Nov 2003 17:02:42 +0200, Dan [EMAIL PROTECTED] wrote:
Hi,
Tried on WinXP Pro and it loads, but in the background (no window).
There is something needed from the wxWindows package to just run the
executable?
Nothing needed from the wxWindows package. I think it's because it
can't
find
the rc directory.
I'm sorry that I didn't put this in the README. Bad coder. No donut.
You must run iaxComm from the installation directory beacuse it looks
for
rc
files in ${cwd}/rc.
Steve put an error dialog on failure in the CVS sources, but I'm
working
on a
better solution.
Please let me know if this solves it, or if the problem lies
elsewhere.
Nope still crashes on XP on load. Ran from directory extracted to, etc.
Below are crash details:
AppName: iaxcomm.exe AppVer: 0.0.0.0 ModName: iaxcomm.exe
ModVer: 0.0.0.0 Offset: 0008e98c
Don't know if that helps any at all, but the other details screen is WAY
too long to attach.
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.538 / Virus Database: 333 - Release Date: 11/10/2003
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Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
Michael, I have the same problem with running iaxcomm. Did the following: * Extract to c:\cd * Open command prompt * c: * cd \cd * iaxcomm What happens: A cursor change for a couple of seconds from an arrow to an arrow with a clock. iaxcomm never shows up. BUT, iaxcomm makes the computer nearly unusable. Killing iaxcomm in the task managers process list helps to make the computer usable again. Where are the configuration information stored? Might it be, that some old configuration information is being used which is no longer of use? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID and CalledID in myapp
Hello.. Does anybody knows how can I use ast_channel-dialed and ast_channel-dialingso I can get callerID and calledID ? There are defined as :ast_channel * dialed ast_channel * dialing and all I need are 2 strings Thanks a lot Alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] g723 to g723 SIP call - warning message
Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [EMAIL PROTECTED] sath]# cat g723.1 - Executing SetCallerID(SIP/-08122ae0, 1001) in new stack -- Executing AbsoluteTimeout(SIP/-08122ae0, 6000) in new stack -- Set Absolute Timeout to 6000 -- Executing Dial(SIP/-08122ae0, Sip/[EMAIL PROTECTED]|90|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnect-c682 is making progress passing it to SIP/-08122ae0 WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames Here is my sip.conf [general] port=5060 context=default allow=g723.1 maxexpirey=180 defaultexpirey=160 ;Connect to iconnect register=1510xx:[EMAIL PROTECTED]/1510xx [iconnect] type=friend secret= username=xxx host=natrelay.deltathree.com dtmfmode=inband canreinvite=no context=vobb-in allow=g723.1 Can someone be able to debug this ? If I make the codec to g729, call not even get through. * complains that it can't bridge the codec. Now in GS phone I can see following setting; Voice Frames per TX: 2(up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) Could there be a mismatch here ? Cheers Sathya ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Don't try to do inland DTMF on anything but G.711. Jeremy McNamara Sathya Weerasooriya wrote: Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [EMAIL PROTECTED] sath]# cat g723.1 - Executing SetCallerID(SIP/-08122ae0, 1001) in new stack -- Executing AbsoluteTimeout(SIP/-08122ae0, 6000) in new stack -- Set Absolute Timeout to 6000 -- Executing Dial(SIP/-08122ae0, Sip/[EMAIL PROTECTED]|90|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnect-c682 is making progress passing it to SIP/-08122ae0 WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect p rocess 1 frames Here is my sip.conf [general] port=5060 context=default allow=g723.1 maxexpirey=180 defaultexpirey=160 ;Connect to iconnect register=1510xx:[EMAIL PROTECTED]/1510xx [iconnect] type=friend secret= username=xxx host=natrelay.deltathree.com dtmfmode=inband canreinvite=no context=vobb-in allow=g723.1 Can someone be able to debug this ? If I make the codec to g729, call not even get through. * complains that it can't bridge the codec. Now in GS phone I can see following setting; Voice Frames per TX: 2(up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) Could there be a mismatch here ? Cheers Sathya ___ 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] TDM400P channel problem
Hi all, I´m trying to set up some analog extensions with two TDM400P cards but I only manage to get the first port/channel working. This works: zaptel.conf fxoks=1 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1 But when I try to set up more ports/channels on the first card it stops working. The following configuration...: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 ...makes * crash with the following message: Ouch ... error while writing audio data: : Broken pipe How will I configure both card to work? Any idea, anyone? ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Jeremy McNamara wrote: Don't try to do inland DTMF on anything but G.711. Er. INBAND Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P driver overwrites memory used bye linux-kernel
Hi, we have a Digium E100P in use with asterisk and the driver of the E100P overwrites important memory locations of the kernel. The side-effects are malloc-errors when simple shell commands are used, unable to compile anything (internal compiler errors), files that are edited are in the state as before editing. The memory is ok, tested this with memtest86 and no problems occur if the driver is not loaded. System: - CPU: Athlon XP 2600+ - RAM: 1 GB DDR400 RAM - BRD: EPOX 8KRA2i KT600 Kernel 2.4.21 (4GB) High Memory Support [*] HIGHMEM I/O support [*] MTRR (Memory Type Range Register) support [*] Local APIC support on uniprocessors [*] IO-APIC support on uniprocessors can the highmem i/o support be the cause of the problems? Has anyone had this problem, too? cu, Steffen Koepf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] creative VoIP blaster *
Ok, I've googled for 15+ minutes, and have yet to find a usable answer, so I'm going to annoy everyone and ask here. I have, in my posession, a creative VoIP blaster. I have installed the fobbit LKM and I can see the device. Can I use it with asterisk in any meaningful way, shape, or form? I'd love to be able to buy an IP phone, ATA, or FXO card, but lack the funds at the moment (won't get into why a grandstream phone at $65 is out of my budget, just take my word for it). Can I turn this hardware that's laying around into anything useful? Thanks, Pat ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/IAX2 DTMF
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP Asterisk2 IAX2 Asterisk1 ZAP PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101. Sending DTMF inband instead of via rfc2833 however eleminates the double DTMF but squelched DTMF is still present in the reverse direction. SIP on Asterisk was still configured as rfc2833 though. In the setup below we do not experience double DTMF from SIP to PSTN but DTMF is still squelched from PSTN to SIP. SIP is configured for rfc2833 on both the endpoint and on Asterisk in this example. SIP Asterisk1 ZAP PSTN It sounds to me like there's a bug here somewhere? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/u1Gi2TEAILET3McRAqzeAJ45ug/n9nMdZPwSq2P9UZl6ontofwCdHMrs MCPHMigosAcVVCr6l+E5lxk= =pj0r -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] creative VoIP blaster *
On Wed, Nov 19, 2003 at 06:07:46AM -0500, Patrick Cantwell wrote: Ok, I've googled for 15+ minutes, and have yet to find a usable answer, so I'm going to annoy everyone and ask here. I have, in my posession, a creative VoIP blaster. I have installed the fobbit LKM and I can see the device. Can I use it with asterisk in any meaningful way, shape, or form? I'd love to be able to buy an IP phone, ATA, or FXO card, but lack the funds at the moment (won't get into why a grandstream phone at $65 is out of my budget, just take my word for it). Can I turn this hardware that's laying around into anything useful? I saw on an Asterisk FAQ somewhere (I too can't find it in Google) but it's here: http://www.asstricks.org/faq.html - 4. I bought a Creative Labs VOIP-Blaster. How can I hook it up to use with Asterisk? You may hook up the VOIP-Blaster to Asterisk, but you can only use it to talk to other VOIP-Blasters. Due to the patent on the codec used by the VOIP-Blaster, it isn't likely that Asterisk will allow the VOIP-Blaster as a simple handset (other than with other VOIP-Blasters) anytime soon. - Anyway, maybe you can find another blaster user, or ask Creative what if you can implement the codec :-) cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to configure G.729 on Asterisk
Hi all, I have an Asterisk installed and I'm using this one only to playback announcements for incoming SIP calls at this moment. These SIP users are registered on another SIP server and not on Asterisk. I'm able to do calls using G711 alaw and ulaw, but I didnt succeed to have something played for G729 or G723.1. I know that I need license for G729, but I saw some emails here that specifies being possible to configure G729 even without license(no one said how to do that). Defintely I'll buy the license for that if the interworking with the SIP server is working. Is here someone that is using currently G729 codec? How is this configured under SIP? Thanks, Constantin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 4 Port FXO cards
Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when?
Re: [Asterisk-Users] 4 Port FXO cards
Yes I am waiting for that card also. Does somebody know when we will be able to buy it? - Original Message - From: Surajee Ratnayake To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 8:51 AM Subject: [Asterisk-Users] 4 Port FXO cards Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when?
Re: [Asterisk-Users] 4 Port FXO cards
On Wed, 2003-11-19 at 05:51, Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? If you spent any time looking over the archive, you should have found that there is a daughter card for the TDM400 card being designed/approved to make FXO ports. This will let you eventually mix and match daughter boards to make the proper mix for your low port density systems. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 4 Port FXO cards
Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P and Meetme
hi all, has anyone come across the following problem? when 1 of 12 zap channels (T100p) in a conference press # a bunch of channels gets kicked off the conference. is this a T100p, zaptel, or meetme problem? are there any solutions? many thanks, patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Jeremy McNamara wrote: Don't try to do inland DTMF on anything but G.711. Jeremy McNamara Someone really needs to patch Asterisk to print some ugly warning or notice to the Asterisk console when the codec that is being used for a call is not ulaw/alaw and trhe dtmfmode=inband (manyually or automagically set) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VOIP onver the net
Hi everybody, I am getting into the * world and I need some help... I woudl like to know the possibilities of doing VOIP over the Internet between 2 or more (point to multipoint) locations I have read documentation from * website, but the examples I came across are usually for 2 points over the PSTN, maybe (quite likely) I have not found the correct documentation, can you please help me ? Thanks in advance !, Sebastian.-
[Asterisk-Users] 2 TE410P
Hi, Is there anybody in this list who had experience with two TE410 cards on a server ? I know that the cards cant share IRQs and Im seeing to have two cards on a x335 IBM Xeon server. TIA -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inter diax connection
hi i am trying DIAX and *. i cannot make calls from ne DIAX to another. whats the config? exten=44,1,Dial(IAX/username,20,tr) ??? just a guess. any help will be appreciated. cm = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax vs iax2 question
Hi!
trunk=yes
and the dialplan includes:
exten = _9.,1,Dial(IAX/npi:[EMAIL PROTECTED]/${EXTEN-1})
I share your confusion about the port setup in iax.conf, but I think you
don't need to worry about iax2 registration - it won't hurt you. Anyway,
from what I understood trunking only works with IAX2 (just a remark).
One final remark: You should really split up the type=friend entry into
two entries for [sai-peer] with type=peer and [sai] with type=user. That
will save you headaches in the future.
Cheers, Philipp
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Re: [Asterisk-Users] Message from * console.
On Wed, 2003-11-19 at 06:10, XISCOAIR wrote: Hi everybody, Now I'm using Asterisk CVS-10/23/03-09:56:06 version, today in my * console has appear a message that I never had seen before. The message is: ' -- Moving call from channel 8 to channel 28' Can someone explain me what it's means?? I have been looking for it but I didn't find it any information. Did you look? How hard is it to do this in the asterisk directory? grep -r 'Moving call from' * This will show you the line in channels/chan_zap.c. Upon looking there I see this is in some PRI functions. Asterisk is well laid out, and it isn't difficult to look up messages like those. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anybody using Sphinx
Anthony Wood wrote: On Wed, Nov 19, 2003 at 10:22:55AM +0800, Steve Underwood wrote: Arnold Ligtvoet wrote: Since I would like the user names to be auto-generated by the system, I would guess that this could best be done using festival with a localized voice. I think there is a Dutch voice for Mbrola with should integrate into festival ( note to self : need bigger harddisk :-) ) Speech recognition accuracy is not great under ideal conditions. Doing what you suggest seems unlikely to achieve any meaningful accuracy. Speech recognition training systems require many occurances of a word or phrase, clearly spoken, before their accuracy becomes useful. A one shot utterance from Festival seems to fail on both counts :-) Sphinx isn't doing general speech recognition, it is determining which of a list of patterns it has you said, like mobile phones do. That is essentially all that any voice recognition currently does. There is little meaningful context directed recognition (a phrase locked loop to use an old in joke) in anything available today. So it's fairly easy to tell between Jennifer and Frank if there are no other options. Many commercial on-line recognisers have serious trouble telling between yes and no when those are the only two acceptable answers. When you call directory assistance in Australia, the IVR asks you what name you want, and gives you a suggestion out of the top 100 or 200 names, which you can accept or reject. Makes for riducule, but beats waiting on hold. Beware that many of these systems are actually a human operator hiding behind and IVR. I've had people tell me about amazing automated directory enquiry systems in the US, which turn out to be a human masquerading as an IVR. If the list is known to be short, that many not be the case here. Bottom line: the very best speech recognition still sucks. As a British speaker I never get more than about 40% accuracy speaking into a US trained recogniser. I have never had better than about 70-80% accuracy on a British trained recogniser. Strangely, my terrible Cantonese gets nearly 100% on SpeechWorks recogniser. :-\ This is true for general speech recognition, where the computer has a much larger dictionary to match the sound waves against. Only a speaker trained system could even begin to approach these accuracies for general text input. The accuracies I gave are for phone based systems expecting a very limited set of responses from an arbitrary caller. Humans really don't do that much better at raw word recognition, but we heavily apply context to improve things. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] inter diax connection
Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 2:41 PM Subject: [Asterisk-Users] inter diax connection hi i am trying DIAX and *. i cannot make calls from ne DIAX to another. whats the config? exten=44,1,Dial(IAX/username,20,tr) The 'username' is registered to * server? How the two DIAX phones are configured? Pls give more details in order to be able to help you... BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Service codes for MGCP channels
Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? Greetings, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] capi config
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rattana BIV Sent: den 18 november 2003 17:56 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] capi config Hi, I have DIVA server BRI with 2 channels and i use chan_capi drivers. But I only can use 1 channel. I make one call it works, but if I make a second call asterisk says me = Everyone is busy at this time. How can I configure it ? Have you tried to put something like: controller=1,2 devices=4 In your capi.conf file? ---JanM--- Best regards Rattana ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax vs iax2 question
Philipp,
trunk=yes
and the dialplan includes:
exten = _9.,1,Dial(IAX/npi:[EMAIL PROTECTED]/${EXTEN-1})
I share your confusion about the port setup in iax.conf, but I think you
don't need to worry about iax2 registration - it won't hurt you. Anyway,
from what I understood trunking only works with IAX2 (just a remark).
I can relate to that, however when a firewall is involved with iax/iax2
connections, one can't generalize or guess at what * might be attempting
without causing service problems, etc. (Installing both is certainly one
way to do it, except security mgmt always involves shutting down ports
that aren't really needed to reduce exposure.)
One final remark: You should really split up the type=friend entry into
two entries for [sai-peer] with type=peer and [sai] with type=user. That
will save you headaches in the future.
I've seen that comment several times before as well. Do you have any idea
or experience as to what headaches might result from its use?
I did have a problem with iaxtel about a month ago and changing to user
seem to correct its stability. But, I really don't know whether it was
a root-cause or symptom.
Can anyone help us better understand user, peer, and friend in the iax
and iax2 environment?
Rich
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Re: [Asterisk-Users] 4 Port FXO cards
anyway, better if Digium can do it quickly, we are suffering a lot with channel banks, we need to replace these channel banks with 4 port cards - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:09 PM Subject: Re: [Asterisk-Users] 4 Port FXO cards Surajee Ratnayake wrote: Hi, Do Digium have any plans to release a 4 port fxo card. If yes, when? I think they are in the pipeline.. Initial speculation was that they would be out in September but I guess there have been problems.. I guess the best answer is they will come out when they come out.. :) 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] Anybody using Sphinx
Bottom line: the very best speech recognition still sucks. As a British ... I don't agree. How many others of you have had the pleasure of saying your account number when calling your credit card company? I have had good success with it, 100% so far. The only engine that I've seen that does suck is when UPS was first using voice responses. I just want to say to keep up the good work. It's a great idea. Many people don't like having to press numbers (our business deals with MANY elderly people). Having them speak a response is a LOT more likely than having them press a number. Imagine: Hi, this is Becky. Would you like to speak with Billing, Appointments, Optical, or a Nurse? . I'm sorry, can you say that again? I didn't understand you. Thank you. I will connect you with Appointments now. Is a lot more friendly than Thank you for calling the Impersonal Office. Press 100 for Billing, 200 for Appointments, 300 for Optical, or 400 for Nurse. If you are so blind that you can't see the numbers on your phone, wait and someone will help you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't connect to digium cvs
i have been in the similar situations... Sometimes you cannot connect to it if it is too busy. good sign isnt it Go get a cup of coffee, try it again you should be able to get it.. Ing. Angel Gomez wrote: Hi all. Is there a problem with digium cvs ? I can't connect to it, it just keeps giving a... cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Coonection refused Thank's ___ 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] RE:inter DIAX connection
Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 4:58 PM Subject: [Asterisk-Users] RE:inter DIAX connection hi, (Dan: i can't paste the conf files as this setup is set in my office, i can paste it tomm if u really need) my conf is this: DIAX -- * -- DIAX/SIP/Zap/etc.. i can call from DIAX to SIP phone, to local calls (using Zap) and international calls using IAX from voicepulse. everything is set ok. i only need to know how connection from DIAX to DIAX or from SIP to DIAX can be made. i can call DIAX to SIP using this line in extensions.conf: exten=44,1,Dial(SIP/myusername,20,tr) or to zap, exten=22,1,Dial(Zap/2,20,tr) any help please. cm I ask you again: Have you register the DIAX phone to Asterisk server? There is an account defined in iax.conf file for the DIAX user? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume X100P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rich Adamson wrote: |Is it enough to just unplug the phones? or do I need to actually remove |all physical connectors/interfaces from the line and run the line |directly from the DMARC to the X101P line in port? | |Don't piss around; connect the X101P directly to the demarc with NOTHING |else connected. If that works, connect the other extensions one at a time |until the problem comes back. | | | Suggest using some very basic troubleshooting techniques... Thanks. | Unplug all the phones; did it help or not? No. It did not. The signal seems to be clearer but still very low. Depending on which level echocancel is set to and whether AGGRESSIVE cancellation is enabled or not on MARK2 determines the level of static present and echo on the VoIP side of the connection, but the caller from a normal analog phone line to the X101P hears none of this. The call is very clear and legible and extremely good (when gnophone going through the VPN doesn't glitch every second or two and cut off what the person was saying; otherwise, what's coming from gnophone is exceptionally good quality). Wonder of wonders, the default settings when setting echocancel=yes (which I read previously on this mailing list is 128) work the best. Setting echocancel to 256 causes some sort of drastic feedback that just about blows the SIP/IAX user's eardrums out. | If not, disconnect the extra wiring; did that help or not? This is going to be more difficult, I think. I would have to undo the wiring from two boxes that the DMARC is split out to inside the building for this particular line. | If you are real sure there is nothing more between your * machine and | the demarc, then call the telco (or your favorite service person that | has the tools and knowledge to actually test what's left). | | It might even be worth your while to simply move the * machine next | to the demarc, and connect it directly to the demarc (with no other cabling | involved whatsoever) and eval the difference. I think I will try this. | Then, to help others understand the issues, report back to the list exactly | what fixed the problem (assuming something realistic solved the problem) | so that others can gain an understanding of unusual things impacting this | stuff. Of course. Thanks for your suggestions. I hope this problem can get licked. Otherwise, we basically just have a voicemail system, and not a very useable one at that. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/u455uYsUrHkpYtARAuVVAJ4s05RzpfocyArHlRJcGEhyuH1GNACdHSfg 1O6ssMuPiiQwLwXKbQ48Njw= =uAbK -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FYI: Simple Small Asterisk install..
Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages individually.. Everything has compiled and Asterisk loads but I haven't tried any Zaptel drivers or hardware so can't comment on that.. Anyway if anyone needs a small secure install there it is.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 TE410P
Hi, I have installed 2 T/E410P in a dual Intel 2.4GHZ HT xeon board 7501HG2 seems stable so far. But I don't have much load at the moment . like 30-40 channels get busy at any time however according to digium 2 cards in such kind of big machine should be okwith full load. Azher"Juan J. Sierralta P." [EMAIL PROTECTED] wrote: Hi,Is there anybody in this list who had experience with two TE410 cardson a server ?I know that the cards can´t share IRQs and I´m seeing to have two cardson a x335 IBM Xeon server.TIA-- Juanjo sin .sig___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard
Re: [Asterisk-Users] iax vs iax2 question
Hi there! One final remark: You should really split up the type=friend entry into two entries for [sai-peer] with type=peer and [sai] with type=user. That will save you headaches in the future. I've seen that comment several times before as well. Do you have any idea or experience as to what headaches might result from its use? [...] Can anyone help us better understand user, peer, and friend in the iax and iax2 environment? See earlier posts of mine here: - type=user cannot register with another server, it must by type=peer (and maybe als type=friend works) - the qualify= setting only makes sense for a peer - the host= setting (plus deny=/permit=) in particular is what can create the unexpected headaches if used with type=friend (some weeks ago there was an excellent posting on this issue, probablly by J Todd), e.g.: you want to fix a host for the peer, but you might not want to fix a host for the user. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to find path from G729A to ULAW on Sipphone.com
Barton/Tilghman, Do I need g729 licenses for Asterisk? I don't really need Asterisk to be a party to the call (i.e. monitor the data stream) and the BT101 apparently includes a license for G.729. Does anybody know if Sipphone/FWD recently changed their system to allow only G.729 calls to traverse the 800 interface? Thanks, Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Hodges Sent: Tuesday, November 18, 2003 3:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Unable to find path from G729A to ULAW on Sipphone.com [EMAIL PROTECTED] wrote: I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A Before somebody tells me UTFG, I ALREADY HAVE. Somebody else had a similar issue last week and there was no real resolution posted. So here it is again. I have all of the codecs that I support enabled in my sip.conf. Here is the relevant section: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm allow=ilbc register = 17476692375:[EMAIL PROTECTED]/1101 [sipphone] type=peer username=17476692375 secret=[MYSECRET] host=proxy01.sipphone.com fromuser=SteveSokol fromdomain=sipphone.com canreinvite=no ; ==END OF SIP.CONF FILE=== The issue occurs whenever any calls that route over the sipphone peer are made to a toll-free number. The calling phone (either my GS100 or my X-LITE softphone) rings two or three times then gives me busy. Here is the entire debug output: -- Executing Dial(SIP/1101-1f83, SIP/[EMAIL PROTECTED]|20|tr) in new stack -- Called [EMAIL PROTECTED] NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1234379840]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A -- SIP/sipphone.com-e7b3 is making progress passing it to SIP/1101-1f83 -- SIP/sipphone.com-e7b3 answered SIP/1101-1f83 -- Attempting native bridge of SIP/1101-1f83 and SIP/sipphone.com-e7b3 NOTICE[1242768320]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1242768320]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) == Spawn extension (default, 918884510851, 1) exited non-zero on 'SIP/1101-1f83' The problem does NOT occur when I call another sipphone.com user (i.e. GS100 - Asterisk - Sipphone - GS100). Those calls go through just fine. The toll free calls were working last week. Is it me, or is it Sipphone.com? Any suggestions would be greatly appreciated. Steve I've been having the same types of problems (I'm probably the guy you're referring to who had the same problems last week). This is the solution I have found to work reliably so far. Configure the Grandstream BT101 with the following codecs, in the following order: choice 1: G.729A/B (g729) choice 2: PCMU (ulaw) choice 3: PCMA (alaw) choice 4: G.729A/B (g729) choice 5: PCMU (ulaw) choice 6: PCMA (alaw) Configure the codecs in sip.conf like this: disallow=all allow=all allow=ulaw allow=alaw allow=g729 Configure the entry in extensions.conf to use a certain codec when necessary (I've found it necessary only when calling through the 800 gateway provided to both FWD and SIPphone): ; FWD exten = _1800NXX,1,Macro(callerid-pstn) exten = _1800NXX,2,SetVar(SIP_CODEC=g729) exten = _1800NXX,3,Dial(SIP/[EMAIL PROTECTED]) ; SIPphone ;exten = _1800NXX,1,Macro(callerid-pstn) ;exten = _1800NXX,2,SetVar(SIP_CODEC=g729) ;exten =
[Asterisk-Users] AMA flags by context
Hi, Is there any way to change the ama flags inside the context, what i want to do is to mark a billing amaflags for outbound trunk contexts ? TIA -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Simple Small Asterisk install..
At 3:41 PM + 11/19/03, WipeOut wrote: Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages individually.. Everything has compiled and Asterisk loads but I haven't tried any Zaptel drivers or hardware so can't comment on that.. Anyway if anyone needs a small secure install there it is.. Later.. I would imagine that if you could get it into 256mb, that would be ideal, since it would fit into many flash disk chips, while 292mb means one would need to buy a 512mb chip (much more expensive.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? Greetings, Philipp No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) http://bugs.digium.com/bug_view_page.php?bug_id=071 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume X100P
| Unplug all the phones; did it help or not? No. It did not. The signal seems to be clearer but still very low. Depending on which level echocancel is set to and whether AGGRESSIVE cancellation is enabled or not on MARK2 determines the level of static present and echo on the VoIP side of the connection, but the caller from If you are fighting only an inbound transmission level problem, then trying changing rxgain=0.0 (or txgain=0.0 for x100p outbound) but, I'm fairly certain a change in those parameters does not take effect until either a complete * restart (or maybe even a complete zaptel/system restart). Someone else on the list might know for sure. Obviously, if you crank the gain up, it will have a degrading effect on the echo canceler as well. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Question on hearing ADSI CAS tone
Hello all, Have a few questions. New to asterisk , just getting setup with 1 X100P and 2 TDM400p. Redhat 9 Hope I sent this to correct list Setting up some Aastra/Vista Powertouch 350 phones. Things work outofbox on ADSI programming, vmail downloads, menus etc. Question. When I Dial voicemail2 mainmenu from keypad on the 350 (Say ext 8) I hear what seems to be the CAS signal from the ADSI transmission. I hear the same thing at the end of the mailbox prompt and end of password prompt. Is it normal to be hearing this sound or should the phone be intercepting it? Prompts and everything works (as much as is programmed) but the sound is annoying as heck. I have noticed I hear this any time the phone is in voice mode, and is trying to switch into data mode for the next set of menu transmissions. I have no alternate equipment or lab or service to comparisons with. First experience with ADSI. So I am not sure if this is normal or not. BTW I also cannot get VMWI working as stated in current bug list. (VMWI broken on TDM400P) Thanks __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume X100P
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rich Adamson wrote: || Unplug all the phones; did it help or not? | |No. It did not. The signal seems to be clearer but still very low. |Depending on which level echocancel is set to and whether AGGRESSIVE |cancellation is enabled or not on MARK2 determines the level of static |present and echo on the VoIP side of the connection, but the caller from | | | If you are fighting only an inbound transmission level problem, then | trying changing | rxgain=0.0 | (or txgain=0.0 for x100p outbound) | but, I'm fairly certain a change in those parameters does not take effect | until either a complete * restart (or maybe even a complete zaptel/system | restart). Someone else on the list might know for sure. | | Obviously, if you crank the gain up, it will have a degrading effect on | the echo canceler as well. Right. The rxgain and txgain are both set to 3.0. It appears that one only needs to shutdown and restart asterisk. At least, I've heard different effects when I do this without unloading and reloading the wcfxo and zaptel modules. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQE/u50uuYsUrHkpYtARAswIAJ4vaojYnARgqhDyvucsS1PB/yR7zACfWcv1 iT+D6rDnBDvAKKmxZHeF2qM= =b4gk -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE:inter DIAX connection
yes, i have successfully registered DIAX with *. i am using diax to call to other phones sip,zap, etc through *. cm --- Dan [EMAIL PROTECTED] wrote: Hi, - Original Message - From: C M [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 4:58 PM Subject: [Asterisk-Users] RE:inter DIAX connection hi, (Dan: i can't paste the conf files as this setup is set in my office, i can paste it tomm if u really need) my conf is this: DIAX -- * -- DIAX/SIP/Zap/etc.. i can call from DIAX to SIP phone, to local calls (using Zap) and international calls using IAX from voicepulse. everything is set ok. i only need to know how connection from DIAX to DIAX or from SIP to DIAX can be made. i can call DIAX to SIP using this line in extensions.conf: exten=44,1,Dial(SIP/myusername,20,tr) or to zap, exten=22,1,Dial(Zap/2,20,tr) any help please. cm I ask you again: Have you register the DIAX phone to Asterisk server? There is an account defined in iax.conf file for the DIAX user? Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't connect to digium cvs
- Original Message - From: Sri [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 9:15 AM Subject: Re: [Asterisk-Users] Can't connect to digium cvs i have been in the similar situations... Sometimes you cannot connect to it if it is too busy. good sign isnt it Go get a cup of coffee, try it again you should be able to get it.. Ing. Angel Gomez wrote: Hi all. Is there a problem with digium cvs ? I can't connect to it, it just keeps giving a... cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Coonection refused Is that handtyped, or is that a typo in a cvs client/server app? Coonection Thank's ___ 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 - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise.Ë^®+$RÇ«²f¢)à+-Ë^®+$RÇ«²X¬¶Çb+¦r¡¶ÚþX¬¶Çb+¦r¿¨¥©ÿ+-wèý«-z¸¬ë ®
[Asterisk-Users] Help please
Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
Hello, John Todd a écrit: Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking ... No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) Do you have any information on how to include these functions to th dialplan? Regards, Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
in my opinion, asterisk is the best solution. it is relatively very cheap, asterisk is free and u can use any soft phones to do calls. you can find cheap digium cards, around $400+. you can set up any normal phone to FXS cards. yu can do conference call and voicemails in asterisk. this link may be helpful to read though there are lots of it out there http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb --- Michael Welter [EMAIL PROTECTED] wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
in my opinion, asterisk is the best solution. it is relatively very cheap, asterisk is free and u can use any soft phones to do calls. you can find cheap digium cards, around $400+. you can set up any normal phone to FXS cards. yu can do conference call and voicemails in asterisk. this link may be helpful to read though there are lots of it out there http://www.voip-info.org/wiki-Asterisk?PHPSESSID=736cb8758a4bc846654fd36e33ead8eb --- Michael Welter [EMAIL PROTECTED] wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO card still won't pick up...
I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial(SIP/3064-564c, Zap/g1/ww954...) in new stack NOTICE[245776]: File app_dial.c, Line 698 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time When I try to call in, usually nothing happens. One time it answered and Asterisk acted like it was going through the dial plan steps, but on the phone I never heard anything. It just hung up immediately. After a couple more seconds Asterisk noticed the hangup and stopped its dial plan. chan_zap.so loads OK, and I can do this: *CLI zap show channels Chan. Num. Extension ContextLanguage MusicOnH 1incoming31 en 2incoming31 en Here's an exceprt from when Asterisk starts: [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Not found (No such file or directory) [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXS Kewlstart signalling DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 2, with 0 conference users -- Registered channel 2, FXS Kewlstart signalling == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' -- Registered to '172.16.0.1', who sees us as 172.16.255.157:4569 WARNING[147466]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable [pbx_config.so] = (Text Extension Configuration) ZTCFG seems OK: # ./ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Zaptel.conf is blank. Zapata.conf follows. These were the same way when the older version was working: # cat zapata.conf [channels] echocancelwhenbridged=yes echocancel=yes stripmsd=1 callerid=asreceived language=en context=incoming3121 signalling=fxs_ks rxgain=3.0 txgain=0.0 usecallerid=yes group=1 channel=1 echocancelwhenbridged=yes echocancel=yes stripmsd=1 usecallerid=no callwaiting=no callerid=intercom 9876543210 context=incoming3130 language=en signalling=fxs_ks group=1 channel=2 So, I'm wondering. 1. What other diagnostic steps can I do to narrow this down? 2. What flags/switches in Asterisk or Zaptel could be causing this, particularly if they've changed recently? 3. Is there a 'how to diagnose zap problems' guide anywhere? Thanks. - Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
On Wed, 19 Nov 2003 09:18:44 +0100, Peer Oliver schmidt [EMAIL PROTECTED] wrote: Michael, I have the same problem with running iaxcomm. Did the following: * Extract to c:\cd * Open command prompt * c: * cd \cd * iaxcomm What happens: A cursor change for a couple of seconds from an arrow to an arrow with a clock. iaxcomm never shows up. I think that I've found the problem. I installed on about every Win32 box I could get my hands on, and could only get two to crash. I think th problem has to do with different display metrics that uncover a wxWindows bug. I've posted a revised binary at http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip Please let me know if this version works for you. BUT, iaxcomm makes the computer nearly unusable. Killing iaxcomm in the task managers process list helps to make the computer usable again. Where are the configuration information stored? Might it be, that some old configuration information is being used which is no longer of use? It's in the registry in \HKEY_CURRENT_USER\Software\iaxComm All of the talk of USB keys has convinced me to (at least optionally) move it to a config file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
On Wed, 2003-11-19 at 10:52, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Thanks for your help, Michael Welter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Updated iaxComm binaries available for WinXP, Red Hat 9.0
Quoting Michael Van Donselaar [EMAIL PROTECTED]: On Wed, 19 Nov 2003 09:18:44 +0100, Peer Oliver schmidt [EMAIL PROTECTED] wrote: Michael, I have the same problem with running iaxcomm. Did the following: * Extract to c:\cd * Open command prompt * c: * cd \cd * iaxcomm What happens: A cursor change for a couple of seconds from an arrow to an arrow with a clock. iaxcomm never shows up. I think that I've found the problem. I installed on about every Win32 box I could get my hands on, and could only get two to crash. I think th problem has to do with different display metrics that uncover a wxWindows bug. I've posted a revised binary at http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-current.zip Please let me know if this version works for you. Michael sent me an updated version (probably this one) and it is working on at least one of the two machines that it failed before on. I have not tried the other system yet. I must admit that I am running at 1152x864 which isn't exactly a standard resolution. Enjoy. BUT, iaxcomm makes the computer nearly unusable. Killing iaxcomm in the task managers process list helps to make the computer usable again. Where are the configuration information stored? Might it be, that some old configuration information is being used which is no longer of use? It's in the registry in \HKEY_CURRENT_USER\Software\iaxComm All of the talk of USB keys has convinced me to (at least optionally) move it to a config file. ___ 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] Service codes for MGCP channels
Citeren Daniel ANDRE [EMAIL PROTECTED]:
Do you have any information on how to include these functions to th
dialplan?
Depends on what you'd want precisely. Here's a (higly undocumented) example
(I'm working on more). I used KPJ's examples on www.junghanns.net to start
with (thanks!) and have a few more tricks up the sleeve that will take more
time to evolve...
Florian
Brief instructions: Make sure the phone has a callerid that matches the
extension number (makes sense, but might not always be true). Then make sure
the phone can dial the entries in [apps].
If you use this setup a phone can dial *21*number for immediate redirect or
*61*number for delayed redirect, and #21# or #61# to cancel the setting.
[macro-stdexten]
;
; Standard extension macro (with call forwarding):
; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten=s,1,DBget(temp=CFIM/${ARG1})
exten=s,2,Dial(Local/[EMAIL PROTECTED]/n) ; Unconditional forward
exten=s,3,Dial(${ARG2},20) ; 20sec timeout
exten=s,4,DBget(temp=CFBS/${ARG1})
exten=s,5,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on busy or unavailable
; No CFIM key
exten=s,102,Goto(s|3)
; No CFBS key - voicemail ?
exten=s,105,Busy
[apps]
; Unconditional Call Forward
exten = _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = _*21*X.,2,Hangup
exten = #21#,1,DBdel(CFIM/${CALLERIDNUM})
exten = #21#,2,Hangup
; Call Forward on Busy or Unavailable
exten = _*61*X.,1,DBput(CFBS/${CALLERIDNUM}=${EXTEN:4})
exten = _*61*X.,2,Hangup
exten = #61#,1,DBdel(CFBS/${CALLERIDNUM})
exten = #61#,2,Hangup
[extensions]
exten=7000,1,Macro(stdexten,7000,MGCP/aaln/1@myphone) ; IP10S
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Re: [Asterisk-Users] Help please
On Wed, 2003-11-19 at 10:52, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are voice mailboxes on each line. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Sorry for the last empty message, don't know if I was able to delete it from my outbox quick enough. Anyways, we discussed pricing recently. Your price for lines is pretty nice. You probably couldn't justify the extra cost of switching to T1 or PRI as the loop itself would cost a little more than you currently are paying. So now you have to figure out how to deal with these lines. You will have to decide whether or not you want IP phones, or analog phones. Either way, you probably are looking at a channel bank and a T1 card to get the 8 lines into a single asterisk machine. 8x12 would fit on a Zhone, but you have to know that the zhone doesn't pass callerid on the FXO ports. You could go with just about any modular channel bank that supports FXO ports. Look at the Adtrans or Adit lines. So for an analog system you are looking at $500 PC $500 T100P card $500 or so for a channel bank of of ebay. $30x12 for phones. or approximately $1860 if you get lucky on ebay. On a purely money view point, if you don't for see growth in the number of lines into your office, you won't see the break even of asterisk for about 4 years over the Centrex lines. Of course you can assign value to having your own machine in house, and the VoIP possibilities, and you can claim break even in a shorter term. Don't take this as downing asterisk, I love it. I just wanted to cut to the dollars since you have placed them on the table. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Simple Small Asterisk install..
Hi, if anyone is looking for a really small * installation take a SuSE rescue system, add sshd, zaptel and *. Gzip it and the image is only 28MB which fits into a 64MB ramdisk and leaves some MBs for temporary logs, CDRs and voicemail. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-11-19 um 16.41 schrieb WipeOut: Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages individually.. Everything has compiled and Asterisk loads but I haven't tried any Zaptel drivers or hardware so can't comment on that.. Anyway if anyone needs a small secure install there it is.. 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] hold music =]
Thanks, I tried this and it doesn't play the sample mp3 file when I place a call on hold. I searched around and found an article (http://www.mail-archive.com/[EMAIL PROTECTED]/msg09971.ht ml) that says to download and install mpg123-0.59q-1.i386.rpm which I did and still no luck... any ideas? Thanks, p.s. I have onboard sound which shows as loaded when I do lsmod Steve. -Original Message- From: Ing. Angel Gomez [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 5:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] hold music =] Steve Bradwell wrote: Hi All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users /usr/src/asterisk/configs/musiconhold.conf.sample and show application setmusiconhold ___ 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] GoTo or Dial in AGI??
I have two possible senarios for making a call from an AGI..
Senario1 - Using GoTo
In the extensions.conf I have..
[dial-out]
exten = _9.,1,AGI(myagi)
exten = _9.,2,Dial(SIP/blah/${EXTEN:1})
In the AGI I have..
EXEC GoTo dial-out|9555678|2
So using this method I don't have to really edit the AGI ever if I
change the dialplan around as long at the context is correct.. At the
end of the AGI it passes the call off to the specified
context/extension/priority.. This way the console reports when the AGI
script has completed and the return code..
Senario2 - Using Dial
In the extensions.conf I have..
[dial-out]
exten = _9.,1,AGI(myagi)
In the AGI I have..
EXEC Dial SIP|blah|9555678
Using this method I don't have to change the dialplan but do have to
edit the AGI script if somthing changes.. The AGI initiates the Dial
command directly.. This way there is no detail that the script has ended
on the console..
So from those who have expereince with AGI, which would you suggest as
the best method to use and why? Have I missed somthing critical that
would make either of these methods a bad idea?
Thanks..
Later..
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Re: [Asterisk-Users] Service codes for MGCP channels
At 5:51 PM +0100 11/19/03, Daniel ANDRE wrote: Hello, John Todd a écrit: Hi there, after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking ... No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) Do you have any information on how to include these functions to th dialplan? Regards, Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com Yes and no. I have a large suite of CLASS features (Vertical Service Codes) integrated into Asterisk, but at this time I am not distributing them publicly. You could, however, build your own on a limited basis using the DBPut and DBGet routines with only a little training with Asterisk's dialplan, or via an AGI. It's just scripting and a few trivial variables about each line that you store and manage. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is how you Search the Archives
Anthony Wood wrote: Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com Does anyone else think there should be a FAQ made specifically for the mailing list, and posted here automatically maybe every 2-4 weeks? -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't connect to digium cvs
Andrew Thompson wrote: - Original Message - From: Sri [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 9:15 AM Subject: Re: [Asterisk-Users] Can't connect to digium cvs i have been in the similar situations... Sometimes you cannot connect to it if it is too busy. good sign isnt it Go get a cup of coffee, try it again you should be able to get it.. Ing. Angel Gomez wrote: Hi all. Is there a problem with digium cvs ? I can't connect to it, it just keeps giving a... cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Coonection refused Is that handtyped, or is that a typo in a cvs client/server app? Coonection handtyped ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
On Wednesday 19 November 2003 10:11, John Todd wrote: after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) You're going to be waiting an awful long time, then, because call features are not going to be removed from the zap channel. Instead, they will be added to more channels. What _will_ be happening in the future, however, is the ability to customize the codes (in both a global, as well as a channel-type-specific way) used to invoke call features. Mark has been very emphatic about call features not belonging in the dialplan. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hold music =]
http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat On Wed, 2003-11-19 at 18:53, Steve Bradwell wrote: Thanks, I tried this and it doesn't play the sample mp3 file when I place a call on hold. I searched around and found an article (http://www.mail-archive.com/[EMAIL PROTECTED]/msg09971.ht ml) that says to download and install mpg123-0.59q-1.i386.rpm which I did and still no luck... any ideas? Thanks, p.s. I have onboard sound which shows as loaded when I do lsmod Steve. -Original Message- From: Ing. Angel Gomez [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 5:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] hold music =] Steve Bradwell wrote: Hi All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users /usr/src/asterisk/configs/musiconhold.conf.sample and show application setmusiconhold ___ 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] FYI: Simple Small Asterisk install..
Klaus-Peter Junghanns wrote: Hi, if anyone is looking for a really small * installation take a SuSE rescue system, add sshd, zaptel and *. Gzip it and the image is only 28MB which fits into a 64MB ramdisk and leaves some MBs for temporary logs, CDRs and voicemail. regards kapejod I know this may seem like a dumb question, how do you run the system from the gzipped file? got any links to info? Guess I need to get to know liinux better.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Hi, Thanks Jeramy and Eric. Sorry for my ignorance. I still did not get the point. Do you mean that I have to set each of my context in sip.conf with dtmfmode=inband ? I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be change to something else ? (Send DTMF:in-audio via RTP (RFC2833) via SIP INFO) Cheers Sathya Date: Wed, 19 Nov 2003 06:15:35 -0600 From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Reply-To: [EMAIL PROTECTED] Jeremy McNamara wrote: Don't try to do inland DTMF on anything but G.711. Jeremy McNamara Someone really needs to patch Asterisk to print some ugly warning or notice to the Asterisk console when the codec that is being used for a call is not ulaw/alaw and trhe dtmfmode=inband (manyually or automagically set) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't connect to digium cvs
cvs [login aborted]: connect to cvs.digium.com(216.207.245.20):2401 failed: Coonection refused Is that handtyped, or is that a typo in a cvs client/server app? Coonection I'll bet you a beer the cvs client uses strerror(errno) to generate the Conection refused string. A quick check with strings `which cvs` | grep refused seems to say I'd likely win the bet. And if I'm correct the error is real as strerror(errno) mostly gets it right. It's hard to know why this would happen without knowing how Digium's server is set up. Is there a firewall, tcp wrappers? is itusing tcpd. who knows??? But from the solaris man page the meaning of Conection refused is The attempt to connect was forcefully rejected. The calling program should close(2) the socket descriptor, and issue another socket(3SOCKET) call to obtain a new descriptor before attempting another connect() call. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is how you Search the Archives
Hi, Citeren Leif Madsen [EMAIL PROTECTED]: Does anyone else think there should be a FAQ made specifically for the mailing list, and posted here automatically maybe every 2-4 weeks? Actually, no. This mailinglist has enough daily traffic to ensure that FAQ would find its way into my trashcan everytime. As an intermediate, one could add a link to the FAQ on the bottom of every message, much like the subscribe/unsubscribe instructions. Then again, nobody seems to read those either :-P... -- Best regards Florian Overkamp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] This is how you Search the Archives
Does anyone else think there should be a FAQ made specifically for the mailing list, and posted here automatically maybe every 2-4 weeks? I was -JUST- thinking that this morning. I've seen this done on other lists, and while it may be a waste of bandwidth... it will most certainly help eliminate quite a few newbie questions. I think this is a great idea. - Devon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
I am sorry I mean dtmfmode=info -Original Message- From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 10:34 AM To: Eric Wieling; [EMAIL PROTECTED] Digium. Com Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Hi, Thanks Jeramy and Eric. Sorry for my ignorance. I still did not get the point. Do you mean that I have to set each of my context in sip.conf with dtmfmode=inband ? I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be change to something else ? (Send DTMF:in-audio via RTP (RFC2833) via SIP INFO) Cheers Sathya Date: Wed, 19 Nov 2003 06:15:35 -0600 From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Reply-To: [EMAIL PROTECTED] Jeremy McNamara wrote: Don't try to do inland DTMF on anything but G.711. Jeremy McNamara Someone really needs to patch Asterisk to print some ugly warning or notice to the Asterisk console when the codec that is being used for a call is not ulaw/alaw and trhe dtmfmode=inband (manyually or automagically set) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RTP timing in a SIP only world (choppy MOH)
I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing? Sip phone to sip phone works fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is how you Search the Archives
- Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 1:04 PM Subject: Re: [Asterisk-Users] This is how you Search the Archives Anthony Wood wrote: Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com Does anyone else think there should be a FAQ made specifically for the mailing list, and posted here automatically maybe every 2-4 weeks? I think it should be individually emailed to the new users with a do you understand query in every known language after it. New users should have to successfully reply to this email before being able to post to the group at large. Also a this is how you unsubscribe test shouldn't be too far behind it :) - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. -- +--+ |Leif Madsen - http://www.hacklocalhost.com| +--+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ,µêâ²E,z»j)b b²Ð,µêâ²E,z»%Ëlv(ºg(m§ÿåËlv(ºg(ùYùb²Ø§~ڲ׫É.±êì
RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Ok, Folks; I set dtmfmode=info on both contexts. So now no warning message, phone rings, but moment I go off hook, asterisk drops the call, and spit out the following; -- Executing SetCallerID(SIP/-081341a8, 1001) in new stack -- Executing AbsoluteTimeout(SIP/-081341a8, 6000) in new stack -- Set Absolute Timeout to 6000 -- Executing Dial(SIP/-081341a8, Sip/[EMAIL PROTECTED]|90|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/iconnect-e3e0 is making progress passing it to SIP/-081341a8 -- SIP/iconnect-e3e0 answered SIP/-081341a8 WARNING[1217602880]: File channel.c, Line 1851 (ast_channel_make_compatible): No path to translate from SIP/-081341a8(4) to SIP/iconnect-e3e0(256) WARNING[1217602880]: File app_dial.c, Line 672 (dial_exec): Had to drop call bec ause I couldn't make SIP/-081341a8 compatible with SIP/iconnect-e3e0 == Spawn extension (vobb-in, 81510xx, 3) exited non-zero on 'SIP/-081341a 8' -- Executing Hangup(SIP/-081341a8, ) in new stack == Spawn extension (vobb-in, h, 1) exited non-zero on 'SIP/-081341a8' It seems like * is trying to translate the codec. I have set G729 for both contexts. I thought if there is no codec translation, asterisk can handle pass through. Cheers Sathya -Original Message- From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 11:01 AM To: Eric Wieling; [EMAIL PROTECTED] Digium. Com Subject: RE: Re: [Asterisk-Users] g723 to g723 SIP call - warning message I am sorry I mean dtmfmode=info -Original Message- From: Sathya Weerasooriya [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 10:34 AM To: Eric Wieling; [EMAIL PROTECTED] Digium. Com Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Hi, Thanks Jeramy and Eric. Sorry for my ignorance. I still did not get the point. Do you mean that I have to set each of my context in sip.conf with dtmfmode=inband ? I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be change to something else ? (Send DTMF:in-audio via RTP (RFC2833) via SIP INFO) Cheers Sathya Date: Wed, 19 Nov 2003 06:15:35 -0600 From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message Reply-To: [EMAIL PROTECTED] Jeremy McNamara wrote: Don't try to do inland DTMF on anything but G.711. Jeremy McNamara Someone really needs to patch Asterisk to print some ugly warning or notice to the Asterisk console when the codec that is being used for a call is not ulaw/alaw and trhe dtmfmode=inband (manyually or automagically set) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can't connect to digium cvs
Done. Thank's all. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is how you Search the Archives
Added to the top of the FAQ. http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] newbie can't start * ---
Thanks, Any help on guidelines for a simple 3 softphones config? Hector. - Original Message - From: Greg Hill [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Martes, 18 de Noviembre de 2003 10:14 p.m. Subject: Re: [Asterisk-Users] newbie can't start * --- I'm not sure about the RH8 system, but I know RH9 comes with some firewall stuff set up (by default?). As root, run 'iptables -L' . If you see something in there about Chain RH-Lokkit-0-50-INPUT or similar (that's the name used on a RH9 box I use), then the machine has some firewalling set up. If you do have firewalling set up, then it's likely that the ports * wants to use are blocked. Scroll back a few hours through the list archives and you'll find a message which mentions what ports * uses.. If you do have firewalls set up like this, then you could entirely remove the firewall by using 'iptables -F; iptables -X' (as root). Again, this completely removes your firewalls, which is probably something you wouldn't want to use for a long-term fix (especially if your machine is visibile on the internet..). There are HOW-TOs and probably other mailing lists where you can learn more about firewalls with iptables. Greg On Tue, 18 Nov 2003, Hcqm wrote: I just compiled * on a redhat 8 system. Make samples also. I don't know much about *, but I started the deamon using -vvvcg and didn't have any error. The console started ok, but trying to reach the * from other machine there is no open ports for connection. Is there anything else I have to do to test it ? I just want to use 3 SIP clients and use * as a PBX with voicemail for a start. I read the hadbook, but can anybody give some additional guidelines or procedure to get it working? Thanks a lot. Hector. ___ 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] Play a sound after dialing a user...
I'd like to play a sound to a user I dial (via SIP) once they answer play the sound and then allow me to talk to them. The new Cisco 7960 SIP code allows to set lines to autoanswer via the speaker phone, I'd like to play a tone after it rings through and then talk... Any thoughts on how to do this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help please
On Wed, Nov 19, 2003 at 09:52:13AM -0700, Michael Welter wrote: Hi, My PBX/key system failed when building power was switched off and then back on. The service rep says the PBX unit is so old that it is not repairable. The unit has 8 incoming POTS lines and 12 multiline sets. There are I have a client running 9 incomming PRI lines with their 768K internet all on one T-1. Works very well. I have another client with 4 pots lines. Works OK, but you have some problems with call detection. voice mailboxes on each line. Voice mail is excellent with Asterisk. Very flexible. Switching to Qwest's Centrex system would cost about $3800/yr. My 8 business lines now cost $3320/yr, so the Centrex increment would be about $480/yr. With Asterisk you would be looking at an initial fixed price investment and about 20% of that per year for support. The cost for pots lines vs. T1 for the PBX would be about the same. You would need to check with LECs and CLECs to determine the most economical way to deliver the PSTN services to your office. Purchasing a new PBX would run anywhere from $4K to $9K. My questions are: should I be considering Asterisk? What type of telephone set could I use with Asterisk? Would I be able to conference outside parties into a call? Would I have voice mail? Absolutely consider Asterisk. You can use standard POTs style phones, some ADSI phones and SIP phones. The Cisco 7960 is very nice. You can set up Meetme conferences and you can conference many incoming calls to the same connection. If you have VoIP channels you could conference them in as well. The client I mentioned above using the T1 PRI does depositions on the conferences. We've had conferences up for 7 hours! Yes, Voicemail is very good and flexible. 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 *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
Florian Overkamp wrote: Citeren Daniel ANDRE [EMAIL PROTECTED]: Do you have any information on how to include these functions to th dialplan? Depends on what you'd want precisely. Here's a (higly undocumented) example (I'm working on more). I used KPJ's examples on www.junghanns.net to start with (thanks!) and have a few more tricks up the sleeve that will take more time to evolve... http://www.voip-info.org/tiki-index.php?page=Asterisk+call+forwarding Looking forward to more of these examples, Florian! I understand I have to find more information on the local channel construct where you use @pbx for some reason I do not understand... /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This is how you Search the Archives
On Wed, 2003-11-19 at 13:15, Andrew Thompson wrote: - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 1:04 PM Subject: Re: [Asterisk-Users] This is how you Search the Archives Anthony Wood wrote: Go to www.google.com type in your search query add this to the end of your search query: site:lists.digium.com Does anyone else think there should be a FAQ made specifically for the mailing list, and posted here automatically maybe every 2-4 weeks? I think it should be individually emailed to the new users with a do you understand query in every known language after it. New users should have to successfully reply to this email before being able to post to the group at large. Also a this is how you unsubscribe test shouldn't be too far behind it :) With the exception of I don't know how hard it is to setup, I wouldn't mind this going to a semi moderated group. RO access requires little intervention. Basically it is the default. Posting requires a quick read of the FAQ with a quick push through a small and to the point netiquete page, and then maybe a 2 or 3 question pop quiz afterwords. After that, release the posting restriction. It is fairly minimalistic, and shouldn't get too in the way of users who want to lurk and read first. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Announced Transfer from Zap to SIP crashes
Hi, If I understand right, From a Zap station, you should be able to flashhook and transfer/3-way a call. It appears to have issues passing calls from PSTN Zap channels - Zap extension - SIP. See below. When Pushing an inbound PSTN Zap call from a zap answering station, and pressing Flashhoook dialing the SIP extension, SIP stations answers, Press flashhook again, we are now on a threeway call, when the Zap user hangs up, * loses it /crashes, with a debug error of File Channel.c Line 2252 (ast_channel_bridge) : Nobody there, continuing. The SIP user is still on the call, the Zap PSTN call is lost, and the Zap answering station is hung up. Is this an issue, or am I going about this the wrong way? Thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hold music =]
Thanks, can this happen from a non - sip phone? We have aastra (bell south) 390 analog phones. -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 1:27 PM To: Asterisk-Users Mailing-list Subject: RE: [Asterisk-Users] hold music =] http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat On Wed, 2003-11-19 at 18:53, Steve Bradwell wrote: Thanks, I tried this and it doesn't play the sample mp3 file when I place a call on hold. I searched around and found an article (http://www.mail-archive.com/[EMAIL PROTECTED]/msg09971.ht ml) that says to download and install mpg123-0.59q-1.i386.rpm which I did and still no luck... any ideas? Thanks, p.s. I have onboard sound which shows as loaded when I do lsmod Steve. -Original Message- From: Ing. Angel Gomez [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 18, 2003 5:02 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] hold music =] Steve Bradwell wrote: Hi All, Just installed our very first asterisk system, and we love it! I cant believe the different things you can do with it, just great =] My question is: How do I configure my system to play an mp3 file when a caller gets put on hold? Thanks in advance, Steve. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users /usr/src/asterisk/configs/musiconhold.conf.sample and show application setmusiconhold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Company
Is there a way to allow someone to hit # for a company directory and step through the extensions? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Simple Small Asterisk install..
try base instalation of debian delete the documentation and asterisk sources should be less than 150megas all Miguel On Wed, 2003-11-19 at 16:12, John Todd wrote: At 3:41 PM + 11/19/03, WipeOut wrote: Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages individually.. Everything has compiled and Asterisk loads but I haven't tried any Zaptel drivers or hardware so can't comment on that.. Anyway if anyone needs a small secure install there it is.. Later.. I would imagine that if you could get it into 256mb, that would be ideal, since it would fit into many flash disk chips, while 292mb means one would need to buy a 512mb chip (much more expensive.) JT ___ 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] RTP timing in a SIP only world (choppy MOH)
On Wed, 2003-11-19 at 16:10, Bob Knight wrote: I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing? Its known problem, Asterisk SIP channels get the timing from the source, so if the source stops transmitting (i.e. VAD) the MoH gets choppy. Try disabling VAD on your Media Gateway. When VAD is active it is usually signaled by an specific RTP payload type, maybe the SIP channel should check that an starts using a local clock. Sip phone to sip phone works fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Company
On Wed, 2003-11-19 at 15:06, Todd Wallace wrote: Is there a way to allow someone to hit # for a company directory and step through the extensions? # can be an extension. exten= #,1,directory -- show application Directory -- -= Info about application 'Directory' =- [Synopsis]: Provide directory of voicemail extensions [Description]: Directory(context): Presents the user with a directory of extensions from which they may select by name. The list of names and extensions is discovered from voicemail.conf. The context argument is required, and specifies the context in which to interpret the extensions. Returns 0 unless the user hangs up. It also sets up the channel on exit to enter the extension the user selected. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
On Wednesday 19 November 2003 10:11, John Todd wrote: after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) You're going to be waiting an awful long time, then, because call features are not going to be removed from the zap channel. Instead, they will be added to more channels. What _will_ be happening in the future, however, is the ability to customize the codes (in both a global, as well as a channel-type-specific way) used to invoke call features. Mark has been very emphatic about call features not belonging in the dialplan. -Tilghman That doesn't make much sense. Reason: call control features need to be managed from the center of the network, not at the edges. I want full control of my end devices, and full visibility into their states. If the CLASS features are handled within the channel, then that implies that either a) a new set of applications or variables exist that can provide visibility and configuration into each channel (yuck!) or b) there is no visibility into set/get on CLASS features (worse.) I have implemented a full CLASS featureset in the dialplan, including customized voice feedback prompts, speed dials, music-on-hold selection, call forwarding (timed AND unconditional), telemarketer block selection, blah blah blah. It took me some time, but it wasn't impossible, obviously - anyone can do it - that's what the dialplan DOES; it's not hardcoded. The concept here is that we want to move the programming into the hands of the admin in a scriptable way, not put the programming inside of the C code of the application package (meaning the chan_* drivers.) Is there a counter-argument to this that addresses the points about visibility into a channel's status? JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Application CallingPres
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+CallingPres Could someone explain this applicatoin a bit more? I found the application in the Zap channel source, and a comment says something about PRI connections. What is the value specifying? Thank you! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application CallingPres
On Wed, 2003-11-19 at 15:21, Olle E. Johansson wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+CallingPres Could someone explain this applicatoin a bit more? I found the application in the Zap channel source, and a comment says something about PRI connections. What is the value specifying? This CVS commit should help. http://lists.digium.com/pipermail/asterisk-cvs/2003-October/000205.html Call presentation is about whether or not callerid is displayed, what callerid is displayed, and similar. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO card still won't pick up...
show channels and zap show channel foo should be helpful. Find me on IRC if worst comes to worst. Mark On Wed, 19 Nov 2003, Matt Lawson wrote: I recently updated (fresh checkout) to the newest zaptel and Asterisk. The one I was using before was a couple of months old. After updating, my zap channels don't work. They won't pick up incoming calls or dial out. When I try to dial out I get: -- Executing Dial(SIP/3064-564c, Zap/g1/ww954...) in new stack NOTICE[245776]: File app_dial.c, Line 698 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time When I try to call in, usually nothing happens. One time it answered and Asterisk acted like it was going through the dial plan steps, but on the phone I never heard anything. It just hung up immediately. After a couple more seconds Asterisk noticed the hangup and stopped its dial plan. chan_zap.so loads OK, and I can do this: *CLI zap show channels Chan. Num. Extension ContextLanguage MusicOnH 1incoming31 en 2incoming31 en Here's an exceprt from when Asterisk starts: [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_oss.so] = (OSS Console Channel Driver) == Console is full duplex == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Not found (No such file or directory) [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXS Kewlstart signalling DEBUG[16384]: File chan_zap.c, Line 1043 (update_conf): Updated conferencing on 2, with 0 conference users -- Registered channel 2, FXS Kewlstart signalling == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' -- Registered to '172.16.0.1', who sees us as 172.16.255.157:4569 WARNING[147466]: File chan_oss.c, Line 238 (sound_thread): Read error on sound device: Resource temporarily unavailable [pbx_config.so] = (Text Extension Configuration) ZTCFG seems OK: # ./ztcfg -vvv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. Zaptel.conf is blank. Zapata.conf follows. These were the same way when the older version was working: # cat zapata.conf [channels] echocancelwhenbridged=yes echocancel=yes stripmsd=1 callerid=asreceived language=en context=incoming3121 signalling=fxs_ks rxgain=3.0 txgain=0.0 usecallerid=yes group=1 channel=1 echocancelwhenbridged=yes echocancel=yes stripmsd=1 usecallerid=no callwaiting=no callerid=intercom 9876543210 context=incoming3130 language=en signalling=fxs_ks group=1 channel=2 So, I'm wondering. 1. What other diagnostic steps can I do to narrow this down? 2. What flags/switches in Asterisk or Zaptel could be causing this, particularly if they've changed recently? 3. Is there a 'how to diagnose zap problems' guide anywhere? Thanks. - Matt ___ 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] RTP timing in a SIP only world (choppy MOH)
Juan, thank you very much. Turning off VAD did it. All is well. Juan J. Sierralta P. wrote: On Wed, 2003-11-19 at 16:10, Bob Knight wrote: I have an * setup with sip phones and sip fxo gateway. When a sip phone places a sip/fxo call on hold, MOH is very choppy. It looks like RTP has a real problem with timing if it is not receiving RTP packets. If the outside call that is placed on hold is not generating any audio, the sip/fxo gateway does not send * RTP packets. Is this valid? Is this a problem with the sip/fxo gateway or a problem with * RTP timing? Its known problem, Asterisk SIP channels get the timing from the source, so if the source stops transmitting (i.e. VAD) the MoH gets choppy. Try disabling VAD on your Media Gateway. When VAD is active it is usually signaled by an specific RTP payload type, maybe the SIP channel should check that an starts using a local clock. Sip phone to sip phone works fine. I connect 2 GS and place one on hold. The GS that is receiving MOH from * is working great because the GS keeps sending back RTP packets. IAX connections work fine. I call an extension on another * box and place it on hold. MOH over IAX/IAX2 is great. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Application CallingPres
On Wed, 2003-11-19 at 15:21, Olle E. Johansson wrote: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+CallingPres Could someone explain this applicatoin a bit more? I found the application in the Zap channel source, and a comment says something about PRI connections. What is the value specifying? This CVS commit should help. http://lists.digium.com/pipermail/asterisk-cvs/2003-October/000205.html Call presentation is about whether or not callerid is displayed, what callerid is displayed, and similar. The value itself is a means of giving information about the callers telephone number. libpri incorrectly processes this as one value, where in actual fact it is two different values, stored within the same byte. The values itself is defined as follows in ITU Q931 Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided In essence, it says 'is the person who has been called allowed to see the callers number' and 'what authority was used to verify that this is a genuine number'. When I have a moment I'll look at contributing a tidyup of the handling of this value as it is something that is important to us as a telco using Asterisk in a carrier environment. Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fritz pci / chan_capi / australia setup
Hi * Fans, I have some fritz cards now, followed instructions from stuart hirsts email of Jun 28: - Thanks for your info but I think I have it working at last. Below are the steps I took which might help others. 1) Download the PCI AVM drivers from ftp://ftp.avm.de/cardware 2) Download the Chan_capi from http://www.junghanns.net/asterisk/ 3) tar -xvzf fcpci-suse8.0-03.09.10.tar.gz which creates a directory called fritz 4) edit the file fritz/src.drv/tools.c and put // in front of the following lines to exclude them : #include linux/string.h #include linux/vmalloc.h #include linux/kernel.h #include linux/version.h //#if LINUX_VERSION_CODE KERNEL_VERSION(2,4,18) #include linux/slab.h //#else //#include linux/malloc.h //#endif This stops the make error message. 5) run make 6) run make install 7) create a file /etc/capi.conf with the following line in it: fcpci - - - - - - 8) run insmod -f fcpci. You will get errors about the version of GCC but it should still work 9) run capiinit 10) run capiinfo which should give output showing the card and supported features. 11) extract using tar -xvzf chan_capi.0.2.2.tar.gz 12) in the chan_capi-0.2.2 directory run make and make install and make config 13) edit /etc/asterisk/modules.conf and add : load = chan_capi.so And in the [global] section add: chan_capi.so=yes 14) then start Asterisk Also forgot to mention that you should make sure that the isdn and hisax modules are loaded by doing : modprobe isdn modprobe hisax I now have incoming calls working OK and working on getting outgoing working. Stuart - OK, so I have some good signs: forge:/etc/asterisk# cat /proc/capi/controllers/1 name fritz-pci io 0xD400 irq 10 type A1 class14 ver_driver 3.09-10 ver_cardtype fritz-pci protocol DSS1 linetype point to multipoint forge:/etc/asterisk# *CLI capi info Contr1: 2 B channels total, 2 B channels free. *CLI I've set up capi.conf as best I can: forge:/etc/asterisk# grep -v ^\; capi.conf | grep -v ^$ [general] nationalprefix=0 internationalprefix=0011 rxgain=0.8 txgain=0.8 [interfaces] msn=292996337 incomingmsn=* controller=1 softdtmf=1 accountcode= context=demo devices=2 forge:/etc/asterisk# And some extensions: grep -i 6337 extensions.conf exten = 11031976,1,Dial,CAPI/292996337:0416059875,5,tr exten = 02111947,1,Dial,CAPI/@292996337:0416059875,5,tr And this is what I get every time: *CLI NOTICE[81926]: File chan_sip.c, Line 5018 (handle_request): Failed to auth enticate user woody sip:[EMAIL PROTECTED];tag=f7f432a8-0b96-4daa-901a-e21e153cf f96 -- Executing Dial(SIP/woody-8607, CAPI/@292996337:0416059875) in new stack -- data = @292996337:0416059875 -- capi request omsn = @292996337 == found capi with omsn = 292996337 == CAPI Call CAPI[contr1/292996337]/1 -- Called @292996337:0416059875 -- CONNECT_CONF ID=001 #0x0005 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 == DISCONNECT_IND PLCI=0x101 REASON=0x3301 -- CAPI Hangingup == No one is available to answer at this time REASON 0x3301 means Layer 1 problem or signalling killed the channel. Is there something I have missed? thanks in Advance -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Service codes for MGCP channels
On Wednesday 19 November 2003 15:16, John Todd wrote: On Wednesday 19 November 2003 10:11, John Todd wrote: after testing with a MGCP phone (Swissvoice ip10s) I found the following ASTERISK-based codes (VERTICAL SERVICE CODES) to work - I assume that most of those will also work with SIP, but haven't checked that yet: *67 - Calling Number Delivery Blocking *70 - Cancel Call Waiting *72 - Call Forwarding Activation *73 - Call Forwarding Deactivation *78 - Do Not Disturb Activation *79 - Do Not Disturb Deactivation *8 - Call pick-up # - Transfer Questions: - how can I enable Calling Number Delivery? *65 doesn't seem to do the trick: - how can I enable Call Waiting after having it disabled via *70? No, these do not work with SIP, and there is currently a request to remove that functionality from the MGCP and Zap channels, since this type of feature should be (IMHO) in the dialplan and not built into the channel. (What if I want to use *70 for something else? How do I read the status of Do Not Disturb, since this is embedded in the channel?) You're going to be waiting an awful long time, then, because call features are not going to be removed from the zap channel. Instead, they will be added to more channels. What _will_ be happening in the future, however, is the ability to customize the codes (in both a global, as well as a channel-type-specific way) used to invoke call features. Mark has been very emphatic about call features not belonging in the dialplan. That doesn't make much sense. Actually, it does. Read on for why I agree with Mark. Reason: call control features need to be managed from the center of the network, not at the edges. And you will have control of your devices. The problem is that many of the call features are intrisically related to how the channel type signals to devices. Hence, the functionality needs to be at the channel driver level. I want full control of my end devices, and full visibility into their states. Putting the functionality inside the channel driver says nothing about visibility. Indeed, one of the things I'd like to do with Zap channel forwarding, for example, is to expose that forwarding with astdb. In addition, this will allow call forwarding on Zap channels to persist across Asterisk restarts. If the CLASS features are handled within the channel, then that implies that either a) a new set of applications or variables exist that can provide visibility and configuration into each channel (yuck!) or That's only if the call features are exposed in the dialplan (which they will not be). Call features will be intrinsic to the channel. b) there is no visibility into set/get on CLASS features (worse.) Again, this is a matter of implementation, not a criticism of the basic concept. I have implemented a full CLASS featureset in the dialplan, including customized voice feedback prompts, speed dials, music-on-hold selection, call forwarding (timed AND unconditional), telemarketer block selection, blah blah blah. It took me some time, but it wasn't impossible, obviously - anyone can do it - that's what the dialplan DOES; it's not hardcoded. The concept here is that we want to move the programming into the hands of the admin in a scriptable way, not put the programming inside of the C code of the application package (meaning the chan_* drivers.) There are two problems, as I see it: first of all, you're cluttering the dialplan with features that ought to be intrinsic to the system. Note that you're going to have to include EVERY context in which phones start with these programmed call features. Second, considering that everybody is going to have pretty much exactly the same logic in every dialplan, that's a lot of wasted time (and a great potential for typos and missed logic). Asterisk is not supposed to be a barebones system (as you seem to be describing); it is, indeed, a full-featured system. But if you really feel strongly about scripting your own call features, note that you can already override existing call features simply by including your own logic for that code in your dialplan. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo cancellation
I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo cancellation
Did you place echocancel=yes before the definition of the channel with channel keyword in zapata.conf ? regards Martin On Wed, 19 Nov 2003, Elijah Chancey wrote: I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey ___ 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] Mediatrix 1102 / 1104 authentication problems....
Hi! Has anyone on the board successfully installed a Mediatrix 1102 or 1104 as a SIP peer on Asterisk? I'm trying to configure different user accounts on each FXS port, but I'm having authentication problems; Asterisk is saying the client is not authorized. Interestingly enough, I can dial a 9 and make a local call through the Mediatrix. Thanks! chris
Re: [Asterisk-Users] echo cancellation
It seems that everyday we see these complaints about bad echo on X100P cards. Why can't these cards incorporate an echo canceller that a cheap $10 dollar phone bought at Walmart can? If we plug a cheap phone on the line there is zero echo. If we plug an X100P on the line there is horrible echoseems to be a daily ocurrence here on the list. Looks to me like analog phones, no matter how cheap, have some sort of effective hardware echo canceller. Why can't an X100P have the same? I would sure like to know. Thanks. - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, November 19, 2003 6:01 PM Subject: Re: [Asterisk-Users] echo cancellation Did you place echocancel=yes before the definition of the channel with channel keyword in zapata.conf ? regards Martin On Wed, 19 Nov 2003, Elijah Chancey wrote: I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey ___ 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] echo cancellation
I am also facing this issue between softphone - softphone and cisco - softphone. I don't have zaptel, so is there any other configuration to apply echo cancellation. Regards, Arslan. -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Thursday, November 20, 2003 4:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] echo cancellation Did you place echocancel=yes before the definition of the channel with channel keyword in zapata.conf ? regards Martin On Wed, 19 Nov 2003, Elijah Chancey wrote: I've got an X100P a cisco 7960. if i call from an analog line via the x100p to the cisco, there is an overly audible echo on the cisco. If i make a call from a cisco to cisco, there is no echo. zapata.conf has echocancel=yes echocancelwhenbridged=yes set. Any ideas? I'm currently using the default implementation of echo cancellation...which one should I try next? elijah chancey ___ 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
