Re: [Asterisk-Users] g723
You must first purchase a license from the G723 patent holders. A license costs about US$10,000. On Tue, 2003-05-27 at 01:52, [EMAIL PROTECTED] wrote: hi! From where do I get the source code for G.723 for asterisk. And how do I compile it (is there any specialities other that make make install )? urs, denzel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Duplicate numbers with outbounding calls
I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose 9 (to do an extern call, see my extensions.conf below), so I've a dial tone. Now I call 0684357917 with my OH323 client but asterisk calls something like 06884335779117 Is it a problem with the french PSTN network ? This are my conf files : - extensions.conf [incoming] ; jouer un fichier des le debut.. exten = s,1,Playback,demo-thanks ;for playing a file ; 9 pour appeler un num exterieur (VoIP-PSTN) ignorepat = 9 exten = 9,1,Dial(Zap/1-1/) exten = 9,2,Congestion - zapate.conf [channels] signalling=fxs_ks context=incoming channel=1 ;X100P ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Duplicate numbers with outbounding calls
Fabrice Tereszkiewicz wrote: I've a problem with my X100P card. I'm setting up a VoIP to PSTN gateway,with oh323. This works, but when I call an PSTN phone number, some digits are duplicated, so I'm unable to call the right person. Not very clear ? I'll try to do better (sorry, I'm french...) example : I use ohphone (with quicknet hardware), I call asterisk (*192*168*1*204#), asterisk answers, I choose 9 (to do an extern call, see my extensions.conf below), so I've a dial tone. Now I call 0684357917 with my OH323 client but asterisk calls something like 06884335779117 Can you verify, from the ohphone console, that the numbers are typed correctly (and not duplicated)? Is it a problem with the french PSTN network ? This are my conf files : - extensions.conf [incoming] ; jouer un fichier des le debut.. exten = s,1,Playback,demo-thanks ;for playing a file ; 9 pour appeler un num exterieur (VoIP-PSTN) ignorepat = 9 exten = 9,1,Dial(Zap/1-1/) exten = 9,2,Congestion - zapate.conf [channels] signalling=fxs_ks context=incoming channel=1 ;X100P Michael. ___ 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] Monitor with Mp3 format
Though I think .mp3 encoding is an interesting idea, even though inefficient at some levels. One benefit is that there are dozens of .mp3 players on the market, with extremely sophisticated library functions. Being able to archive calls in one of the .mp3 library programs might be of some use. However, .wav is almost always playable by those same programs, so perhaps only .gsm is problematic (sorry, not a Windoze expert, don't know the apps there very well.) Another possible use for encoding into .mp3 would be to send the output to a pipe, which would in turn connect to a streaming server such as shoutcast or icecast, which would in turn then allow lots of standard applications to re-broadcast the audio stream. The lack of an paging (and intercom) feature most SIP phones is a serious, serious problem in the business world and this might offer another tool to partially get around that shortcoming. JT You can not record directly to mp3 anywhere in asterisk. It is not something that would be truly wanted if you realized how much overhead it would cause and how little you would gain from a gsm file. On Tue, 2003-05-27 at 05:23, Thomas Haeger wrote: Hi all, i want to monitor channels with the Monitor app in mp3 formatted files. But, if i use mp3 as first argument for the Monitor app, errors occurs WANRING[28689]: File file.c, Line 193 (ast_writestream): Unable to translate to format mp3, source format 64 ...following more Warnings . Is it not possible to monitor to mp3-files? What's wrong? Thanks for Help, Thomas. [snip] -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] The Phantom Call.. T1 card too
I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten = s,1,Hangup (I'm just using the line for outbound dialing) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Levente Sent: Tuesday, May 27, 2003 11:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Phantom Call.. Same thing happened with me too. X100P. Same US tones Sometimes it gets into the voicemail too:)) And the voicemail record 3 minutes tone, after 1.5minutes it's service not available or something similar. Is there a fix for that? - Original Message - From: Mark Street [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 27, 2003 5:39 PM Subject: Re: [Asterisk-Users] The Phantom Call.. Funny, I just noticed this happening on my box with 2 X101P's installed and a phone connected to the same line as one of the X101P's. I pick up the phone after 1 ring, or call someone. After a minute or two * picks up the line and starts the greeting. I pull the plug on the asterisk box to continue the conversation. I just noticed it happening a couple of weeks ago. US dialtone here... On Tuesday 27 May 2003 08:13, Mark Spencer wrote: Could it be that the X100P is detecting the UK dial tone as a ring?? or Has anyone else had a similar problem when using the X100P/S100U combination?? It's possible there is *something* on the line that is confusing Asterisk into thinking a ring takes place. You might try adjusting the value of PEGCOUNT in wcfxo.c to a higher value (say, 10). -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc ___ 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] Monitor with Mp3 format
On Tue, 2003-05-27 at 11:24, John Todd wrote: Though I think .mp3 encoding is an interesting idea, even though inefficient at some levels. One benefit is that there are dozens of .mp3 players on the market, with extremely sophisticated library functions. Being able to archive calls in one of the .mp3 library programs might be of some use. However, .wav is almost always playable by those same programs, so perhaps only .gsm is problematic (sorry, not a Windoze expert, don't know the apps there very well.) .gsm files can be converted quickly after the fact with sox. Sox should just put the wav header on the front of the file that identifies the codec and chunk sizes, then does the bit shifting to make a valid MSGSM file. GSM codec is available on all windows systems by default. Another possible use for encoding into .mp3 would be to send the output to a pipe, which would in turn connect to a streaming server such as shoutcast or icecast, which would in turn then allow lots of standard applications to re-broadcast the audio stream. The lack of an paging (and intercom) feature most SIP phones is a serious, serious problem in the business world and this might offer another tool to partially get around that shortcoming. If you wanted to pipe to a streaming server, you would probably pipe the raw audio somewhere then have a encoder listening on that. It is more inline with the way streaming is done for live streams. You can not record directly to mp3 anywhere in asterisk. It is not something that would be truly wanted if you realized how much overhead it would cause and how little you would gain from a gsm file. On Tue, 2003-05-27 at 05:23, Thomas Haeger wrote: Hi all, i want to monitor channels with the Monitor app in mp3 formatted files. But, if i use mp3 as first argument for the Monitor app, errors occurs WANRING[28689]: File file.c, Line 193 (ast_writestream): Unable to translate to format mp3, source format 64 ...following more Warnings . Is it not possible to monitor to mp3-files? What's wrong? Thanks for Help, Thomas. [snip] -- Steven Critchfield [EMAIL PROTECTED] ___ 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] The Phantom Call.. T1 card too
Sorry, I seem to have responded to the wrong thread there. Should have been part of the Duplicate numbers thread. On Tue, 2003-05-27 at 12:36, Joe Antkowiak wrote: No popping/bad audio on this one, clear as can be, asterisk just decides to pick up the channel after about a minute and use the s extension in the context... immediate=no is set on this channel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, May 27, 2003 1:36 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] The Phantom Call.. T1 card too I had a similar problem when there was timing problems with my T100P. You could also hear lots of popping and generally bad audio during the dial tone. After we fixed the timing problem, the audio was clear as could be, and the problems went away. Of course this doesn't fix a X100P as it is strictly analog. On Tue, 2003-05-27 at 12:03, Joe Antkowiak wrote: I've had the same thing happen, only on the single port T1 card and a channel bank, and one of the FXO channels also having a phone attached elsewhere... I just wound up putting that channel in a different context and running Exten = s,1,Hangup (I'm just using the line for outbound dialing) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tamas Levente Sent: Tuesday, May 27, 2003 11:45 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Phantom Call.. Same thing happened with me too. X100P. Same US tones Sometimes it gets into the voicemail too:)) And the voicemail record 3 minutes tone, after 1.5minutes it's service not available or something similar. Is there a fix for that? - Original Message - From: Mark Street [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 27, 2003 5:39 PM Subject: Re: [Asterisk-Users] The Phantom Call.. Funny, I just noticed this happening on my box with 2 X101P's installed and a phone connected to the same line as one of the X101P's. I pick up the phone after 1 ring, or call someone. After a minute or two * picks up the line and starts the greeting. I pull the plug on the asterisk box to continue the conversation. I just noticed it happening a couple of weeks ago. US dialtone here... On Tuesday 27 May 2003 08:13, Mark Spencer wrote: Could it be that the X100P is detecting the UK dial tone as a ring?? or Has anyone else had a similar problem when using the X100P/S100U combination?? It's possible there is *something* on the line that is confusing Asterisk into thinking a ring takes place. You might try adjusting the value of PEGCOUNT in wcfxo.c to a higher value (say, 10). -- Mark Street, D.C. Red Hat Certified Engineer Cert# 807302251406074 -- Key fingerprint = 3949 39E4 6317 7C3C 023E 2B1F 6FB3 06E7 D109 56C0 GPG key http://www.streetchiro.com/pubkey.asc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Detail Record Analysis Packages?
Can anyone share any links regarding packages to do Call Detail Record (CDR) analysis from the CDR Master file? Login-distance reconciliation, billback, and data presentation are three primary areas of interest. Thanks in advance for your help! --Nick -- Nick Eggleston Consultant Data Communications Consulting, Inc. 6320 Rucker Road, Suite E Indianapolis, IN 46220 317/726-0295 x18 317/202-2445 (fax) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM400P BT
I have to get some ModTaps (converters) too. My question, is do they need to be Master, Slave or PBX type ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: 27 May 2003 16:21 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TDM400P BT I haven't used it but I am sure the US to UK converter thats available from maplin (www.maplin.co.uk) should work.. From the site under Communications Telephones Telephone accessories BT to RJ45 Adapters - 3 Products from £4.99 to £6.99 BT to US Extension Cords - 3 Products from £3.99 to £5.99 BT/US Adapters - 2 Products from £4.99 to £6.99 Hope that helps.. Hi folks, Any Brits on the list got experience of hooking devices with standard BT plugs to the TDM400P? The TDM400P has the RJ11/RJ45 socket but I'm not sure if a standard adapter will do the trick or if we need to consider ring capacitors etc. Any links to buy the stuff would also be appreciated. Cheers, Simon -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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] Kernel Version for CAPI AVM Fritz PCI V2 / chan_capi / chan_alsa update to latest version..
Hello there I have a serious issue with the AVM Fritz PCI V2 I have the following setup and the problem is, that the kernel freezes hard after about 16 hours. The second problem is, that the S-Bus gets jammed as well, so you can't even use a analog phone! on the NT Kernel 2.4.21rc2 with ACPI Patch and of course capi are there any reasons why this configuration should not work? And the second thing is could any of the skilled C programers get the chan_alsa to work with the latest alsa version , I tried really hard to do it myself but my C is simply not sufficient :-(.. I think the problem has something to do with the buffer readout because I am able to get the playback working, but not the recording so the switching/mixing of those 2 is not really ok. And btw. the last working alsa-version is 0.9.0rc7 which is dated January 2002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
On Tue, 27 May 2003 11:48:19 -0700 (PDT), Brad Bergman wrote: I think SayDigits will say anything for which there is a sound file in the digits directory. So if you put a S.gsm file there, SayDigits,S98 should say Star Nine Eight. I realize that's not exactly what you're looking for. Close, and you are right, anything in the digit directory will work as such, problem i have is * how the heck can you have a asterisk(star).gsm file ??? I notice you use S but thats not really a star. Basically Our system (emulating a normal telco here) uses the star key as a set key and a hash key as a terminator on a string of digits or separator. The hash now works, but the start I need a reback method ( festival sounds terrible) On Tue, 27 May 2003, Gary wrote: Any chance of say digits being extended to recognise * # ?? Heck these are digits on a normal keypad :-) Gary . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk PostgreSQL
On Tue, 2003-05-27 at 18:51, Steve Bourg wrote: I noticed someone mentioning PGSQL in this mailing list. I found an older message about it being used for CDR. Is there any info about the extent to which Asterisk and PostgreSQL can interact? I see that there is an Application in the source. Am I restricted to what I can code in C? pgsql is for connecting to postgres from the extention logic. I use it to look up our oncall user and call them. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
Ah, now has anyone got a gsm of thevoice for start and hash ?? On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ 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] chan_h323 + Ericsson Webswitch 100
Turn on h.323 debug and then possibly h.323 trace 4 (for the hardcore) in the Asterisk CLI Unfortunately a tcpdump tells me nothing. Jeremy McNamara Nick wrote: I'm haveing trouble connecting an Ericsson Webswitch 100 to asterisk. Has anyone gotten a Webswitch running? When I try to connect asterisk thinks everything works fine, while the webswitch just rings. I belive chan_h323 is picking the wrong port to talk at the webswitch on, however I'm not sure, nor am I sure how to fix it. Any clues/hints? A tcpdump is attached to show the session. Thanks Nick 16:36:32.116059 192.168.101.253.1719 lpr-2.pelagiris.org.1719: udp 121 16:36:32.120109 lpr-2.pelagiris.org.1719 192.168.101.253.1719: udp 33 (DF) 16:36:32.310700 192.168.101.253.1069 lpr-2.pelagiris.org.1720: S 218957757:218957757(0) win 8192 mss 512,nop,wscale 0,nop,nop,timestamp 19012 0 16:36:32.310758 lpr-2.pelagiris.org.1720 192.168.101.253.1069: S 538397878:538397878(0) ack 218957758 win 5792 mss 1460,nop,nop,timestamp 154312862 19012,nop,wscale 0 (DF) 16:36:32.373145 192.168.101.253.1069 lpr-2.pelagiris.org.1720: . ack 1 win 8192 16:36:32.594192 192.168.101.253.1069 lpr-2.pelagiris.org.1720: P 1:270(269) ack 1 win 8192 16:36:32.594222 lpr-2.pelagiris.org.1720 192.168.101.253.1069: . ack 270 win 6500 nop,nop,timestamp 154312891 19012 (DF) DEBUG[426004]: File chan_h323.c, Line 960 (setup_incoming_call): Sending Ericsson to context [default] 16:36:32.600461 lpr-2.pelagiris.org.1720 192.168.101.253.1069: P 1:155(154) ack 270 win 6500 nop,nop,timestamp 154312891 19012 (DF) 16:36:32.624619 lpr-2.pelagiris.org.1720 192.168.101.253.1069: P 155:361(206) ack 270 win 6500 nop,nop,timestamp 154312894 19012 (DF) 16:36:32.681821 192.168.101.253.1069 lpr-2.pelagiris.org.1720: . ack 155 win 8042 16:36:32.740325 192.168.101.253.1069 lpr-2.pelagiris.org.1720: . ack 361 win 7990 DEBUG[442389]: File rtp.c, Line 787 (ast_rtp_write): Ooh, format changed from 0 to 2 16:36:33.622651 lpr-2.pelagiris.org.1720 192.168.101.253.1069: P 361:546(185) ack 270 win 6500 nop,nop,timestamp 154312994 19012 (DF) 16:36:33.624943 lpr-2.pelagiris.org.61886 192.168.101.253.30006: udp 45 (DF) 16:36:33.651466 lpr-2.pelagiris.org.61886 192.168.101.253.30006: udp 45 (DF) 16:36:33.671474 lpr-2.pelagiris.org.61886 192.168.101.253.30006: udp 45 (DF) 16:36:33.691474 lpr-2.pelagiris.org.61886 192.168.101.253.30006: udp 45 (DF) ___ 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] SayDigits
Heck, I am not that fussy !! Actually, if we could actually get festivel to be fully understandable and using thevoice I think we could all be a lot happier :-) On Tue, 27 May 2003 20:52:58 -0400, Richard Alexander wrote: I suspect that the (American) voice would have called the hash pound in any case.. :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Sent: Tuesday, May 27, 2003 8:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SayDigits Ah, now has anyone got a gsm of thevoice for start and hash ?? On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ we actually can record to file *.gsm #.gsm so thats for that tip. . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bindaddr and multiple ethernet connections
Title: Message My asterisk machine has a dual port adaptec ethernetcard in it, one port configured with a public IP and the other with a private (10.0.0.5). To get my Cisco 7960 phone to sucessfully registerwith asterisk, I have to configure bindaddr to 10.0.0.5 and not the default 0.0.0.0. How can I configure the system to also accept SIP connections on the public IP? Thanks, Marcus
[Asterisk-Users] Wildcard X100P x NTT FSK
With a X100P, I would be able to get the user's caller id here in Japan? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bridging two outbound calls with iconnecthere
Hi All, Is is possible to bridge two outbound calls via iconnecthere, so that two people can be connected. If so what do I need to put in the extensions.conf. Thanks for all the help. Pradeep _ Himalayan holiday! Waiting to be won. http://server1.msn.co.in/sp03/summerfun/index.asp Find out more. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller-ID questions and suggestions
1. It would be very helpful if the voicemail module included the caller-ID info (name and number) in the Subject line. Otherwise, looking at the list of messages is like looking at an INBOX without a From column. I patched app_voicemail.c quickly, but it would be even better if this were a configuration option (since some installations don't have caller-ID information available, and ...from unknown would just be a waste of time). 2. When calls come in to an ATA-186 with a garden-variety caller-ID display attached, the date and time always show up as 01/01 12:00. I am not sure whether this is because nothing is getting sent, because a mis-formatted date is getting sent, or because the ATA-186 is incapable of sending the time/date to the display. In either case it's annoying for me since I use that as my desk clock! Any solutions would be very welcome. 3. Caller-ID name information is sent with double quotes around it, at least when it's fetched from the database using LookupCIDName. This wastes precious display space. 4. There seems to be no way to distinguish between Out of Area and Private/Withheld in extensions.conf. I see in the zaptel source code that the distinction is available when the data is being read from the line. Since those really are different situations, it would be great to be able to differentiate between them. miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID questions and suggestions
There seems to be no way to distinguish between Out of Area and Private/Withheld in extensions.conf. I see in the zaptel source code that the distinction is available when the data is being read from the line. Since those really are different situations, it would be great to be able to differentiate between them. I noticed this in the source code also, but I have observed that calls are always displayed (on a telephone plugged into an FXS port) as Unknown Name/Number, even when Private Name/Number is received from the PSTN, and that internal calls, if *67'ed, display as Unknown rather than Private. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo cancellation
Hi Klaus, Yes - el-cheapo BRI means the cheapest ISDN BRI card I could find at the local hardware pusher, absolutely no-name locally (Malaysia) manufactured Hisax type 20 card from isdn4linux point of view and yes - chan_modem_i4l. I primarily bought the card just to check out how Asterisk perform with such a card and generally it's ok actually. Just that annoying echo thingy. I don't think the Snom Phones can adjust the rx/tx gain in any way. I don't question your knowledge, but just out of curiousity - can you explain why I only see this problem when I receive a call on the Snom's - not when I originate a call? Any ideas? Any settings that can be modified in Asterisk? Regards, Lars -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Klaus-Peter Junghanns Sent: 27 May 2003 20:02 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo cancellation hi Lars, does el-cheapo BRI mean chan_modem_i4l? the echo you hear is caused by todays very sensitive pstn phones. the mic picks up the sound from the speaker and sends it back to you. try to reduce the rx and tx volumes of your snoms (if that is possible), or add rx/tx gain support to chan_modem_i4l like i have for chan_capi. 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 Die, 2003-05-27 um 14.49 schrieb Lars Boegild Thomsen: Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN network to any of the Snom phones, the user on the Snom phone is hearing a bad echo of himself. This echo is however NOT heard on the PSTN side. Any hints? Regards, Lars... -- Lars Boegild Thomsen Technical Director JustIT Sdn. Bhd. Cell Phone (MY): +60 (16) 323 1999 ICQ: 6478559 Yahoo Chat: [EMAIL PROTECTED] MSN Chat: [EMAIL PROTECTED] http://www.justit.ws ___ 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] 2 4-port T1 cards
3. Dual MB won't help much in pure telephony. In pure telephony, you are basically dealing with serial line IO. A T1 is little more than I long distance serial line. 8 T1s is just 11.7megs per second each way, or 23.4 megs in and out. Not too much for a good machine to do. Granted, if you are doingVoIP then you add another set of ins and outs with compression in the middle of it too. This is where the second CPU comes in handy. Actually, with Zaptel, and the T400P especially, dual CPU makes a *big* difference. The T400P and E400P are slave-only designs, so the CPU spends a lot of time just cramming I/O down the PCI bus. Having a second CPU free to do work will definitely help. 4. AGP Video. Make sure not to use the frame buffer, it has been reported thatthe frame buffer generates large amounts of interupts and will degrade the performance. Don't underestimate this effect or think that a fast CPU will get around it. frame buffer is a definite no-no because it disables interrupts during screen redraws which take an enormous amount of time. If your call quality drops while you're playing quake on your PBX, don't come crying to us ;-) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller-ID questions and suggestions
On Tue, 2003-05-27 at 22:52, Miguel Cruz wrote: 2. When calls come in to an ATA-186 with a garden-variety caller-ID display attached, the date and time always show up as 01/01 12:00. I am not sure whether this is because nothing is getting sent, because a mis-formatted date is getting sent, or because the ATA-186 is incapable of sending the time/date to the display. In either case it's annoying for me since I use that as my desk clock! Any solutions would be very welcome. Just a thought, but did you provide the ata186 with a ntp server? 3. Caller-ID name information is sent with double quotes around it, at least when it's fetched from the database using LookupCIDName. This wastes precious display space. Did you define them with double quotes? You may want to go looking into the db and see if it has double quotes. If so you have your answer. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for call logging.......?
Hi All, I wonder if Asterisk can be setup as a Call Centre and do Call Logging? I want to log all incoming and outgoing calls. What sort of logging mechanisms does it supports? Any help is greatly appreciated. Thanks in advance! Regards, Denzel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Conferencing
Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, No ISA Tormeta card found. Does conferencing require some special hardware on the machine on which * is running ?? thanks rahul __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX and Data support in asterisk......?
greetings! We are planning to deploy Asterisk PABXs between several offices. The connection between the offices are going to be on IAX channels. Currently we use G.729 for the IAX channel. But for sending FAXes between offices over IP, this method is not gonna be satisfactory since with this codec there is a data loss. Is there any other lossless codec that we could use to send FAX over IP ? Ideally we would like to have G.729 reserverd for voice calls and a lossless codec for FAX service. Can anybody suggest a way to do this ? - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, May 27, 2003 6:54 PM Subject: Re: [Asterisk-Users] FAX and Data support in asterisk..? On Tue, 2003-05-27 at 07:10, [EMAIL PROTECTED] wrote: Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send text/images to another phone of the same kind, can they sit on FXS extension of Asterisk and do the same thing? If yes, what's the codec that needs to use? Also can we set it up to support one codec for Voice and another codec for Data? If you are using non VoIP connections between telephony hardware, you don't have to worry about codecs. Codec choice only comes into play if you are doing some VoIP connections where bandwidth is of concern. As far as phone connections and fax connections, again if you are using hardware from Digium, you just plug them in just like any other phone and they just work. You only run into trouble if you are doing VoIP over much of a distance. While I have had some luck getting a 14.4 connection over my cable modem using Zap hardware, it isn't a good idea to do it for important calls. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
