Re: [Asterisk-Users] G726-16 passthrough...
I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't recognize the payload tpyes for G726-16. Asterisk does not support G726-16. It only supports G726-32. 1.0.x only supports 32kbps, but I beleive HEAD supports more than -32. roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agents priority in queue
Hello Ppl. Please share info how have you set Agent priority in one queue. Or there is no such kind of thing in current version ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Feedback on CBMySql, MeetMe2 and web interface
Actually I'm using mysql. I couldn't get it to compile w/o the pq libs though so I had to install postgress. It's connecting to mysql, but I do get this stuff in the console: Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1426 get_db_params: PostgreSQL database port not specified. Using default 5432. Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1441 get_db_params: PostgreSQL database table not specified. Using default meetme_user. Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1456 get_db_params: PostgreSQL database sequence not specified. Using default id_meetme_user. Mar 23 03:43:16 WARNING[6356]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! [app_cbmysql.so] = (Conference Bridge MySQL) Kris Dan Austin wrote: Just a guess, but you are using Postgres? When I started working on/with the MeetMe2 gui I saw the same problem, found in the archives that others were seeing it and that using MySQL just worked. I tested with Postgres and confirmed that sql updates were not being written back to the database, but queries worked fine. It might be a permissions problem or a malformed query. Fixing it is next on my list, since I think I want to use a few of the more advanced Postgres features to extend the functionality of the interface. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards Sent: Tuesday, March 22, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Feedback on CBMySql, MeetMe2 and web interface Hi Dan, This sorta works for me. The only thing that doesn't work is the actual admin functions (changing mode of users from Listen to Listen and Talk, or Kicking users). I can see whose in the conferece and see if they are a user or admin though. kRis Dan Austin wrote: I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have the apps doing most of what we want and need. About a week ago I uploaded a new tar.gz of the web interface that added sort and page breaks to the Update/Delete conference listings. My next update will integrate the participants caller ID into the Monitor component, but that is my last planned non-bug fix update. If there is something else the wbe interface should provide, I would love to hear about it. If no one has been able to make it work, well that would be good feedback too. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BV Outbound Drop fixed .
Compiled from CVS today and no more dropping outbound calls after 40 secs. :D (was using cvs from 3 days ago) (just thought I'd pass it along in case anyone is still strugling with broadvoice calls) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I change the volume on a sip phone (Snom) from *?
I have some Snom 190's but the volume from is really low (speaker is ok). Is there any way this can be changed on the server? On the Snom I already set the volume to maximum. Thanks! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Hello Bashir, what kind of problems are you having with oh323 ? George Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Features/Dial Codes (Newbie question?)
I just installed [EMAIL PROTECTED] and connected it to broadvoice via SIP. I've setup an auto-attendant, extensions and everything it working extremely well. I can dial in, I can dial out. Everything works great. What I am using is as follows: Box is an generic Intel box with a PIII/1Ghz processor and 256Mb of RAM and a 4 Gb drive. Network is behind a basic NAT from a Linksys cable router. Outbound is provided by broadvoice.com VoIP provider via SIP. I'm using X-lite softphones from xten. I have found everything pretty much self-explanator and easy to use. Ican not, however, despite a lot of searching on the internet and a lot of reading and checking pages, find answers to a few basic questions. My questions are as follows: 1. I've searched and searched. I've read the entire draft of the users guide. I can not find a decent list of the dial codes, such as this short list: *72Call Forwarding System *73Disable Call Forwarding *77IVR Recording *78Enable Do-Not-Disturb *79Disable Do-Not-Disturb *90Call Forward on Busy *91Disable Call Forward on Busy *98Enter Message Center (also found that *98exten enters extention directly) *99Playback IVR Recording Simulate incoming call 1234System will tell you your extension 2. I want to know if there is a dial code or a way of putting calls on hold. I've set this up toget some experience with Asterisk to evaluation a possible migration from our Inter-Tel system at work to asterisk. It is on my system at home, but if I can not get a hold feature, it would not work for work. I got all of this working after reading an article from a link on slashdot.org to http://www.geekgazette.com/index.php?option=com_contenttask=viewid=2Itemid=26and it's all working great. I want to test the on hold system and the on hold music, but I can not. I appreciate your answers. GP No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.0 - Release Date: 3/21/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP: Failed start after install asterisk_oh323-0.7.1
Hello Charles, due to the recent changes made in asterisk CVS asterisk_oh323-0.7.1 is not up to date yet. It will be in the next few days. George Charles Wang wrote: Hi, ALL: I install my oh323 channel driver following steps of http://www.oinko.net/astrecipes/index.php?action=artikelcat=270174id=10artlang=en I works my asterisk well before install the chan_oh323.so. But after I do make install the oh_323, my asterisk crash and show me the following message (asterisk -vvc). Does anyone have any idea about it? What's wrong about ir? -- Error Message -- [chan_oh323.so] = (InAccess Networks OpenH323 Channel Driver) Mar 21 11:13:25 WARNING[16199]: config_old.c:27 ast_load: ast_load is deprecated, use ast_config_load instead! == Parsing '/etc/asterisk/rtp.conf': Found Mar 21 11:13:25 WARNING[16199]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! == Parsing '/etc/asterisk/oh323.conf': Found [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6 [1]WrapGatekeeperServer::WrapGatekeeperServer: Creating new gatekeeper. Ouch ... error while writing audio data: : Broken pipe Segmentation fault oh323.conf [general] listenAddress=myip listenPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=5 libTraceFile=/var/log/asterisk/oh323.log gatekeeper=mygnugk ;gatekeeperPassword=secret gatekeeperTTL=600 userInputMode=TONE amaFlags=default accountCode=myaccount context=voip-h323 [register] alias=h323248 alias=248 [codecs] codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax receive issues and NVFaxDetect
From: Chris Tuska [EMAIL PROTECTED]
Subject: [Asterisk-Users] Fax receive issues and NVFaxDetect
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,7,rxfax(${FAXFILE})
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(7)
[from-Sipmedia2]
;second line in or Fax line
exten = s,1,Answer
exten = fax,2,Goto(fax,2901,1)
[fax]
exten = 2901,1,Macro(faxreceive)
exten = h,1,System(/var/lib/asterisk/scripts/mailfax ${FAXFILE}
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
I'm a little bit confused here. Assuming the from-Sipmedia2 context is
handling your SIP appliance, how about the following (just moved around a
little of yours and added a line)?
[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = s,2,rxfax(${FAXFILE})
exten = s,3,System(/var/lib/asterisk/scripts/mailfax ${FAXFILE}
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[from-Sipmedia2]
exten = s,1,Answer
; You need this line in here
exten = s,2,NVFaxDetect
exten = fax,1,Goto(fax,2901,1)
[fax]
exten = 2901,1,Macro(faxreceive)
Jusitn Newman
[EMAIL PROTECTED]
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[Asterisk-Users] Some audio problems
Hi all. I have a problem to hear one side, when the second is working fine. softphone - ser - asterisk (IVR) - extension in IVR - ser - pstn - regular phone. The audio which coming from regular phone i can hear without problem, but the audio from softphone i can not hear in the regular phone. here is the log what i am receiving: 9 headers, 9 linesFound RTP audio format 8Found RTP audio format 101Peer audio RTP is at port xxx.xxx.xxx.xxx:27232Found description format telephone-eventCapabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)set_destination: Parsing sip:[EMAIL PROTECTED];ftag=as4783926c;lr=on for address/port to send toset_destination: set destination to serserverip, port 5060 inside sip.conf disallow=all allow=ulawallow=alaw now my soft phone using G729,G723,alaw Any help will be more than appreciated. Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] features enableing via database per extension number
I am looking for a way to add features to an extension number. e.g. extension 601 gets features a, b and c, while extension 605 gets the features a, d and e. I would like at the beginning query a database to get the flags for the extension (bellow for 601) feature_a=y feature_b=y feature_c=y feature_d=n feature_e=n How to ask a database ??? I found a ifgoto, but not an ifinclude or do I miss it somewhere? Has somebody done something like that bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Feedback on CBMySql, MeetMe2 and web interface
The warnings are semi-bogus, since the dbtype can be either MySQL or Postgress. Beyond the fact that the warnings refer to Postgres when using MySQL, it appears that these config Options are not set. port= sequence= dbtable= Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards Sent: Wednesday, March 23, 2005 12:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Feedback on CBMySql, MeetMe2 and web interface Actually I'm using mysql. I couldn't get it to compile w/o the pq libs though so I had to install postgress. It's connecting to mysql, but I do get this stuff in the console: Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1426 get_db_params: PostgreSQL database port not specified. Using default 5432. Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1441 get_db_params: PostgreSQL database table not specified. Using default meetme_user. Mar 23 03:43:16 WARNING[6356]: app_meetme2.c:1456 get_db_params: PostgreSQL database sequence not specified. Using default id_meetme_user. Mar 23 03:43:16 WARNING[6356]: config_old.c:39 ast_destroy: ast_destroy is deprecated, use ast_config_destroy instead! [app_cbmysql.so] = (Conference Bridge MySQL) Kris Dan Austin wrote: Just a guess, but you are using Postgres? When I started working on/with the MeetMe2 gui I saw the same problem, found in the archives that others were seeing it and that using MySQL just worked. I tested with Postgres and confirmed that sql updates were not being written back to the database, but queries worked fine. It might be a permissions problem or a malformed query. Fixing it is next on my list, since I think I want to use a few of the more advanced Postgres features to extend the functionality of the interface. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Edwards Sent: Tuesday, March 22, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Feedback on CBMySql, MeetMe2 and web interface Hi Dan, This sorta works for me. The only thing that doesn't work is the actual admin functions (changing mode of users from Listen to Listen and Talk, or Kicking users). I can see whose in the conferece and see if they are a user or admin though. kRis Dan Austin wrote: I've had 50+ people download the web components, and other than reports of compile issues, I have not heard if this collection has worked for anyone. I do plan to keep updating the * applications and the web pages, but I have almost meet all of our internal requirements and wonder if anyone else is finding it usefull. My focus has been and will likely stay on the user interface, since I have the apps doing most of what we want and need. About a week ago I uploaded a new tar.gz of the web interface that added sort and page breaks to the Update/Delete conference listings. My next update will integrate the participants caller ID into the Monitor component, but that is my last planned non-bug fix update. If there is something else the wbe interface should provide, I would love to hear about it. If no one has been able to make it work, well that would be good feedback too. Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blog post on Asterisk setup
Many friends of mine asked me to describe my home Asterisk setup. I've done that at: http://www.rfc1149.net/blog/index.php/mrhyde/2005/03/23/asterisk_build_your_own_pbx (or use the shorter http://tinyurl.com/5s79m URL) Sam -- Samuel Tardieu -- [EMAIL PROTECTED] -- http://www.rfc1149.net/sam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk locking up - 99.9% CPU
On Wednesday 23 March 2005 05:27, Peter Illmayer wrote: Hello We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work with our call agent. Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9% CPU. There is no debug output or other information that indicates there is a problem... Rather than continually restarting, can anyone make suggestions as to how we can track this down **OR** has anyone got the latest oh323/pwlb to work with CVS Head ? Use top to find the offending process and not its Process ID (PID) - goto super user mode and use the strace command : strace -p pid This should give you a log of the calls being made and should reveal any loops that * may have got itself into. Use this info to visit the source code. It's a start... PaulH -- Paul Hewlett (Linux #359543) Email:`echo [EMAIL PROTECTED] | rev` Tel: +27 21 852 8812 Cel: +27 72 719 2725 Fax: +27 86 672 0563 -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXS FXO
Hi, How do I connect two Analog PBX together with Asterisk.I want two simultaneous voice channels between sites. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXS FXO
to connect to an extension line on you analogue pbx you need an FXO card in your asterisk box. If your analogue pbx has ISDN capabilities then you may be able to connect the two together via an additional isdn card in your analogue pbx and asterix box and a x over cable which would be a far better solution. Chris - Original Message - From: Michael Sanders [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 11:30 AM Subject: [Asterisk-Users] FXS FXO Hi, How do I connect two Analog PBX together with Asterisk.I want two simultaneous voice channels between sites. Thanks Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IRQ headaches
Gary,
I am using TE110P card with [EMAIL PROTECTED] an I had some trouble to setup
corectly, maybe my experiance helps you
Excuse my ignorance here, but I am desperately trying to isolate the IRQ
for
my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled
all
TE110P card should use wcte11xp drivers I am not sure about t1xxp (maybe
someone know if this is same under /proc/interrupts)
1. Edit /etc/init.d/zaptel and add driver for TE110p, and removed ztdummy
from there
MODULES=torisa tor2 wct4xxp wct1xxp wcte11xp wcfxo wcfxs wcusb
RMODULES=wcusb wcfxs wcfxo wcte11xp wct1xxp wct4xxp tor2 torisa
2. Edit /usr/src/zaptel/wcte11xp.c and add some lines to look like:
static struct pci_device_id t1xxp_pci_tbl[] = {
{ 0xe159, 0x0001, 0x71fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79fe, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x795e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x79de, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0xe159, 0x0001, 0x797e, PCI_ANY_ID, 0, 0, (unsigned long)
Digium Wildcard TE110P T1/E1 Board },
{ 0 }
};
3. cd /usr/src/zaptel; make clean; make install
for this asterisk should be down (I suppouse)
4. /etc/init.d/zaptel restart
I am running on an HP Compaq D530s with Fedora Core 1, here is my
I use CentOS (it should be similar)
Mario
[EMAIL PROTECTED]
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Re: [Asterisk-Users] why even use SIP
5) MWI, Call Waiting, 3-way calling missing If I remember correctly (only used an IAXy a couple of times), it uses shutter-tone to tell you when there's a message waiting It definitely support Call Waiting : just use Flash as with normal call waiting on the PSTN Never tried 3-way calling, but I think it supports it 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Let's face facts there, the IAXy sucks by any definition. No it doesn't. Granted it has a couple shortcomings, but nothing that bad. If Digium can fix the most important ones and find a way to drop the price a bit, this would be a great little device. Just my $0.02 CDN ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playback of sound files but no sound
Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') Are the sound drivers (alsa or oss) used for this or do I need to configure something else? thanks for any help, Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback of sound files but no sound
The most common cause for this is there being no timing source available. Do you have the zaptel drivers correctly installed and configured? You could just enable 'ztdummy' and test the system using that. On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote: Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') Are the sound drivers (alsa or oss) used for this or do I need to configure something else? thanks for any help, Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ext matching problems
Hello, sorry for this very late answer, I check the my inbox almost 4
times per hour but never saw the answer. Mind you it has almost 3 days
there :P.
Ok, 'cuz i've been playing around I just changed the sip channels' names
so now instead of shipchan1001 and sipchan1002 they are just the ext
number 1001 and 1002, but the dialplan is exactly the same.
here's the output when I press 0 to from any of the phones:
*CLI Setting NAT on RTP to 0
Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found
Setting NAT on RTP to 0
Check for res for 1001
Call from user '1001' is 1 out of 0
build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=1
-- Executing Answer(SIP/1001-2def, ) in new stack
-- Executing Playback(SIP/1001-2def, fcopba1) in new stack
Ooh, format changed from unknown to ulaw
-- Playing 'fcopba1' (language 'en')
Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found
-- Executing Hangup(SIP/1001-2def, ) in new stack
== Spawn extension (default, 0, 3) exited non-zero on 'SIP/1001-2def'
-- Executing Answer(SIP/1001-2def, ) in new stack
-- Executing Playback(SIP/1001-2def, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Executing Playback(SIP/1001-2def, goodbye) in new stack
-- Playing 'goodbye' (language 'en')
-- Executing Hangup(SIP/1001-2def, ) in new stack
== Spawn extension (default, h, 4) exited non-zero on 'SIP/1001-2def'
update_user_counter(1001) - decrement inUse counter
Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found
---
And this one is when I press * to call the vm form any phone and
introduce the password (no matter if get in the mailbox or not when the
vm-machine stops answering my call the pasvalide context plays on.):
*CLI Setting NAT on RTP to 0
Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found
Setting NAT on RTP to 0
Check for res for 1001
Call from user '1001' is 1 out of 0
build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;line=1
-- Executing VoiceMailMain(SIP/1001-7bab, 1001) in new stack
Ooh, format changed from unknown to ulaw
-- Playing 'vm-password' (language 'en')
Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found
Sending dtmf: 57 (9), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
-- Incorrect password '9' for user '1001' (context = any)
Difference is 9096, ms is 1157
-- Playing 'vm-incorrect' (language 'en')
-- Playing 'vm-password' (language 'en')
Sending dtmf: 56 (8), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
-- Incorrect password '8' for user '1001' (context = any)
Difference is 4184, ms is 543
-- Playing 'vm-incorrect' (language 'en')
-- Playing 'vm-password' (language 'en')
Sending dtmf: 55 (7), at 192.168.0.65
Sending dtmf: 35 (#), at 192.168.0.65
-- Incorrect password '7' for user '1001' (context = any)
Difference is 4848, ms is 626
-- Playing 'vm-incorrect' (language 'en')
-- Playing 'vm-goodbye' (language 'en')
Locked path ''
Unlocked path ''
-- Executing Hangup(SIP/1001-7bab, ) in new stack
== Spawn extension (default, *, 2) exited non-zero on 'SIP/1001-7bab'
-- Executing Answer(SIP/1001-7bab, ) in new stack
-- Executing Playback(SIP/1001-7bab, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Executing Playback(SIP/1001-7bab, goodbye) in new stack
-- Playing 'goodbye' (language 'en')
-- Executing Hangup(SIP/1001-7bab, ) in new stack
== Spawn extension (default, h, 4) exited non-zero on 'SIP/1001-7bab'
update_user_counter(1001) - decrement inUse counter
Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found
---
Any idea???
Francisco.
P.S.: sorry I'm answering to everybody so this messages hits your
mailbox directly, it's that I took so long to answer so I'm not sure if
you alredy forgot the thread :P. Very sorry indeed. Not gonna happen
again.
Le lundi 21 mars 2005 à 16:03 -0500, C F a écrit :
What is your CLI output?
On Mon, 21 Mar 2005 15:03:14 -0400, Francisco Moreno
[EMAIL PROTECTED] wrote:
Hello everyone...
I'm trying to get up a testing pbx installation. Following instructions
of what've read from the handbook and from asterisk's wiki, I wrote the
dial plan as follows:
[general]
;
;
static = yes
;[globals]
;
[default]
;
exten = 0,1,Answer()
exten = 0,2,Playback(fcopba1)
exten = 0,3,Hangup()
exten = *0,1,Answer()
exten = *0,2,Record(fcopba1:gsm)
exten = *0,3,Playback(fcopba1)
exten = *0,4,Hangup()
include = extentions
include = pasvalide
[extentions]
exten = 1001,1,Dial(SIP/sipchan1001,10)
exten = 1001,2,Voicemail(u1001)
exten = 1001,3,Hangup()
exten = 1002,1,Dial(SIP/sipchan1002,10)
exten = 1002,2,Voicemail(u1002)
exten = 1002,3,Hangup()
exten = *,1,VoicemailMain(${CALLERIDNUM})
;exten = *,1,VoicemailMain()
exten = *,2,Hangup()
[pasvalide]
exten = _.,1,Answer()
exten = _.,2,Playback(invalid)
exten =
[Asterisk-Users] Any Software Echo Cancellation in Asterisk?
Hi, short question: Is Echo Cancellation in Asterisk somewhere done by software or is it exlusive done by hardware? Christian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Channels
In a old mailing list, someone got the trouble, anyone has idea? I am getting this when I do a: show sip channels 209.82.xxx.xxx 0071495217 2591218534@ 00103/1 unknow(d) 209.82.xxx.xxx 0041590104 0690231739@ 00103/1 unknow(d) 209.82.xxx.xxx 0070259259 3265102826@ 00103/1 unknow(d) 209.82.xxx.xxx 0071948143 1927207026@ 00103/1 unknow(d) 209.82.xxx.xxx 0022576786 1752809624@ 00103/1 unknow(d) 209.82.xxx.xxx 0070153955 0085223171@ 00103/1 unknow(d) I have about 60 of them and growing. I have submitted a ticket with my provider to let them know of this problem but I would like to clear them out w/o restarting the asterisk binary. thanks gupiter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any Software Echo Cancellation in Asterisk?
-Original Message- From: Christian Gerstner [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 23, 2005 4:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Any Software Echo Cancellation in Asterisk? Hi, short question: Is Echo Cancellation in Asterisk somewhere done by software or is it exlusive done by hardware? Software - see the file mec2.h in the zaptel source code for the most commonly deployed echo can. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] snom 220 version
Good day all What is a good stable snom 220 firmware version. Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Reg Asterisk
hi, Is asterisk a registrar server. thanks, satish Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or [EMAIL PROTECTED] immediately and destroy all copies of this message and any attachments. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium support quality: Excellent
On Tue, Mar 22, 2005 at 05:58:47PM -0500, [EMAIL PROTECTED] wrote: I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me alternatives to hopefully correct my problem! And she was patient and friendly! I nearly fell off my chair. If you have any doubts about buying Digium products, don't let lack of support stop you. They stand behind their products with both technical support and customer service. You can't really ask for more than that. I agree that they are eager to correct any issues that we have with the cards. The unfortunate thing is that the TDMs are so problematic. I'm not sure if it's due to inconsistencies in the hardware into which they are put or the cards themselves or what. I have not yet successfully put 2 TDM cards into a system (though I know others have) and I recenly had a problem where loading the TDM driver and starting * would cause the outgoing message to be played way too fast. I was told to try changing PCI slots (I haven't had a chance to do that yet), but since the TDM cannot share IRQs with anything else, changing slots might just put it into a conflict situation. This one could be sticky... -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reg Asterisk
[EMAIL PROTECTED] wrote: hi, Is asterisk a registrar server. It all depends. If you mean registrar for Inter-Galaxy Travel Permissions, no. If you mean SIP registrar, yes. But we are not a SIP proxy ;-) /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] read dtmf during dial
hello list; Is there a way of catching dtmf during a call ? I tried to use an AGI, which launch the Dial command, and a command like WAIT FOR DIGIT in a parallel thread, but asterisk don't give any response until the dial is hangup. I also thought using a MeetMe app, connecting the two peers in the same conference room and running an agi in the background, but I read on the wiki that it won't run with the SIP protocol (the only one I use :)) Can someone give me a clue ? merci d'avance thibault. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] who has purchased a V400 card from Varion ?
Does Varion not provide any support for their products? I'm interested to know why you chose them over Digium... Andrew, We are a small shop that had one t100p card, when it came time to expand to a second card we found the price had been raised to $599 from $499 for the single port. The 4 port cards from varion are $699 on special. I believe that I can get it working without any support from varion. It would be irresponsible of me to buy a 1 port for the office at that price. Plus I think 1 card instead of 2 would be a better solution in my server. I support Digium any way I can ( t100p, plus 2 TDM cards/x100-non clone when I was first starting out last year) but $100 buys me 3 more ports. If I'm wrong I'll be return the card to varion and buying the new TE110p. Ernest Stokes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom 220 version
Hi, I had some issues especially with voicemail with the latest stable version. I took the chance to run the beta version snom200-SIP 3.56m. So far so good, it solved the vmail issues (dialing vmail from the message button failed). And I have not noticed any strange behaviour sofar. Max. Op wo, 23-03-2005 te 14:32 +0200, schreef Altus Snyman: Good day all What is a good stable snom 220 firmware version. Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GR-303 from Central Office supported?
I'm a little confused on whether the GR303 support in * will accept calls from a Siemens central office that has GR303. Anyone know for sure? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADIT 600 Dynamic Impedance matching
Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are far less however we're still getting them. While I have not been witness to this myself it still seems to be a problem. I understand the quality of wire plays a big part and yes speaker phones and cell phones can attribute to these problems remote or local. The only clue to the dynamic impedance is that the 5g and ver8 of the FXS cards can hardcode the impedance according to country. Well that's fine and dandy but so can a Rhino CB-24 in the rating of milliamps.. Does anyone have suggestions regarding these issues? Please hold back the flaming comments. I'm not here to flame, but to resolve and very tiring issue. :-) Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice alternatives
Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback of sound files but no sound
Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? thanks again, Bart -Original Message- From: Gareth Blades [mailto:[EMAIL PROTECTED] Sent: woensdag 23 maart 2005 13:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback of sound files but no sound The most common cause for this is there being no timing source available. Do you have the zaptel drivers correctly installed and configured? You could just enable 'ztdummy' and test the system using that. On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote: Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') Are the sound drivers (alsa or oss) used for this or do I need to configure something else? thanks for any help, Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisk as VMail server on CCM 3.3.3 System
I've started to look at how I might go about using the Voicemail functionality of Asterisk in our Call Manager environment we have in the office. We're currently using Unity but want to look at moving off that if at all possible. I've read a few things on the wiki and also taken a look through the mailing list to see what info is out there to do this kind of setup, but also wanted to put a call out to anyone who may have accomplished this kind of setup and would be willing to share some info. From what I've read so far, a H323 trunk between * and CCM is the way to interconnect the two systems. There's been mention of having a H323 Gatekeeper in the mix somewhere but I haven't had much chance to take a look at this further on how this might hang together. We're not really looking to upgrade to CCM4 in the near future so SIP Trunks are kind of out of the Q. If anyone has any suggestions on where I might start reading up, or can pass on their experience I'd be very much appreciative. Would the setup I've mentioned with a H323 trunk be considered as ready for use in production today? TIA Nathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback of sound files but no sound
Bart Van Daal wrote: Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? That's because you do NOT need a Zaptel timer for Playback, Background, or MoH. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * and Cisco Callmanager Interconnection
Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP callid
Hello all, I tried the dev list, but got no answer at all. I'm facing some problems with call-id generation in a heavily loaded Asterisk Server. Asterisk is generating same call-id and from tag for different calls (and this is not desirable). Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? Maybe my system lacks of a good random number generator. Is that possible? What is necessary for a linux box (Debian, in my case) to achieve good random numbers (and consequently good callids)? Best Regards, Chuck. __ Do you Yahoo!? Yahoo! Mail - Find what you need with new enhanced search. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback of sound files but no sound
That's because someone suggested it earlier on the list. So I installed the ztdummy driver. Could you please tell me what is needed to playback sound files? thanks, Bart -Original Message- From: Eric Wieling aka ManxPower [mailto:[EMAIL PROTECTED] Sent: woensdag 23 maart 2005 14:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback of sound files but no sound Bart Van Daal wrote: Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? That's because you do NOT need a Zaptel timer for Playback, Background, or MoH. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k is installed I have a x100p card and it doesn't detect a hangup from the calling party when going in voicemail(). My PSTN provider is sending open loop disconnect (voltage decrease for a given moment of time). Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not be required to fix this problem. I wonder if disconnect supervision is the same as open loop disconnection. Actually, the service is ok from my POTS provider. I did put an AbsoluteTimeout on the voicemail() but is not a viable solution as for if somebody wants to leave a 10mn message, he wont be able to do so. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
Vicky Shrestha wrote: Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. I just signed up with Broadvoice myself and got it to work just last night. I followed the directions here: http://www.geekgazette.com/index.php?option=com_contenttask=viewid=2Itemid=26 http://www.geekgazette.com/index.php?option=com_contenttask=viewid=20Itemid=26 The one lingering issue I'm having is choppy sound. JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home print incoming fax
try it out and tell us if it works. That would be a
cool feature.
--- Tim Litwiller [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] has this for it's incoming fax macro
--- start snip ---
[ext-fax]
exten = in_fax,1,GotoIf($[${FAX_RX} =
system]?2:analog_fax,1)
exten = in_fax,2,Macro(faxreceive)
exten = in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11
${FAXFILE} | ps2pdf
${FAXFILE}.pdf)
exten = in_fax,4,system(mime-construct --to
${EMAILADDR} --subject Fax
from ${CALLERIDNUM} ${CALLERIDNAME} --attachment
${CALLERIDNUM}.pdf
--type application/pdf --file ${FAXFILE}.pdf)
exten = in_fax,5,system(rm ${FAXFILE}
${FAXFILE}.pdf)
exten = in_fax,6,Hangup
exten = analog_fax,1,GotoIf($[${FAX_RX} =
disabled]?3:2) ;if fax is
disabled, just hang up
exten = analog_fax,2,Dial(${FAX_RX},20,d)
exten = analog_fax,3,Hangup
;exten = out_fax,1,wait(7)
exten = out_fax,1,txfax(${TXFAX_NAME}|caller)
exten = out_fax,2,Hangup
exten = h,1,Hangup()
--- end snip ---
If I just want it to print to my printer I should be
able to setup a
printer and the use
exten = in_fax,3,system(lpr 11 ${FAXFILE})
and then renumber 5 6 to 4 5 or is this not
possible?
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[Asterisk-Users] Diva Server configuration
Hello Can someone tell me how do I configure a Eicon Diva Server BRI with Asterisk? Should I use CAPI? And how do I tell Asterisk to use QSIG? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback of sound files but no sound
Bart Van Daal wrote: That's because someone suggested it earlier on the list. So I installed the ztdummy driver. Could you please tell me what is needed to playback sound files? Nothing. It Just Works. You call into your Asterisk server, dial the extension for the Playback or Background, or whatever and hear the file being played. Simple. Something ELSE is going on here. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] why even use SIP
On Wed, 23 Mar 2005 06:54:03 -0500, Time Bandit [EMAIL PROTECTED] wrote: 6) Configuration requires Linux, as opposed to a web browser or something more standard. I compiled iaxprov on Cygwin, works nicely. There's somebody on this list that made a Windows version to provision it. Works nicely, GUI interface, can even scan the LAN to find IAXy. Here is the link to it : http://dacosta.dynip.com/asterisk Thanks for that link, I'm gonna try it. The main issue here is that this is a large company and I don't have access to the DHCP servers, and therefore can't just find out the IP address of this thing. There's another feature request. Let me dial ### or something to find my IP... Let's face facts there, the IAXy sucks by any definition. No it doesn't. Granted it has a couple shortcomings, but nothing that bad. If Digium can fix the most important ones and find a way to drop the price a bit, this would be a great little device. You just admitted that without the features, it's not great... Seems like we're all on the same page here. No sense arguing about it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
I have a x100p card and it doesn't detect a hangup from the calling party when going in voicemail(). My PSTN provider is sending open loop disconnect (voltage decrease for a given moment of time). Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not be required to fix this problem. I wonder if disconnect supervision is the same as open loop disconnection. Actually, the service is ok from my POTS provider. I did put an AbsoluteTimeout on the voicemail() but is not a viable solution as for if somebody wants to leave a 10mn message, he wont be able to do so. You might want to look at voicemail.conf and configure the maxsilence for ten seconds or so. That doesn't address your disconnect supervision, but does work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming response and external access
Asterisk only uses UDP, and AFAIK, you also need UDP ports 1 to xx see /etc/asterisk/sip.conf for details. You will also need to set the various NAT related config options in the sip.cfg file. As far as getting it to work on your LAN, well, I though you said you had X-Lite working for internal calls, which implies it is working on the LAN at least ??? What are the procedures required to a. Call my Asterisk box from eyeBeam/Ineen b. Connect to my Asterisk box as a proxy from eyeBeam/Ineen I setup IPCop to only allow my home (static) IP as a security measure. I can call extension on the LAN fine, and call external numbers through the modem, it received phone calls from the phone line no problem, just the External Internet to Asterisk that the problem. Hi Paul, where are you from? We're probably related. :) Chances of that, Smith, Jones, King maybe but not Goodyear :) and a Asterisk user too. I'm from the UK up t'North. Is there the same latency when you configure x-lite directly to your VoIP provider? We dont have a VoIP provider!!! Just the Asterisk box conencted to phone line. Get in to what? You mean SIP to your machine from outside? For IAX2, the manager interface and so on, there are another 8 or so ports you need open and forwarded. As I said above really, to either connect to my Asterisk box as a proxy, or just call it from eyeBeam/Ineen. Thanks All. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback of sound files but no sound
Do you have these files :- [EMAIL PROTECTED] default]# ls -l /dev/zap total 0 crws-T 1 root root 196, 254 Mar 22 09:21 channel crws-T 1 root root 196, 0 Mar 22 09:21 ctl crws-T 1 root root 196, 255 Mar 22 09:21 pseudo crws-T 1 root root 196, 253 Mar 22 09:21 timer If you are using udev you need to add some configuration manually so these files get created. There is a note displayed when making the zaptel package but it is easy to miss. On Wed, 2005-03-23 at 13:40, Bart Van Daal wrote: Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? thanks again, Bart -Original Message- From: Gareth Blades [mailto:[EMAIL PROTECTED] Sent: woensdag 23 maart 2005 13:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback of sound files but no sound The most common cause for this is there being no timing source available. Do you have the zaptel drivers correctly installed and configured? You could just enable 'ztdummy' and test the system using that. On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote: Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround(SIP/joa-9def, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') Are the sound drivers (alsa or oss) used for this or do I need to configure something else? thanks for any help, Bart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADIT 600 Dynamic Impedance matching
On March 23, 2005 08:25 am, Matt Schulte wrote: Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We That's odd, I have always had excellent support from CAC. And FWIW I've never had echo problems with their channel banks. Ever. I have echocancel turned off in the Zapata driver. The only clue to the dynamic impedance is that the 5g and ver8 of the FXS cards can hardcode the impedance according to country. Well that's fine and dandy but so can a Rhino CB-24 in the rating of milliamps.. You don't tune impedance in milliAmps. That's a current measurement. The Rhino can probably alter the amount of current it can source and this is what they're talking about. Not having used Rhino's stuff, I can't say for certain, but you simply don't alter impedance by changing mA. (yes, IAAEE). Does anyone have suggestions regarding these issues? Please hold back the flaming comments. I'm not here to flame, but to resolve and very tiring issue. :-) You can start by giving us a connection diagram between the Adit600 and whatever you're hooked up to, including grade of cable, how long it is, what it's terminating to (make and model) and whether you've tried replacing some runs with other cable to test. Invariably my Adit600 analogue runs are always under 50 feet since I'm terminating to a PBX or KSU nearby. These devices are able to terminate very long (km) runs, so I am curious as to why you're having such issues. Do you have the gains on the Adit600 or Zapata turned way up? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playback of sound files but no sound
On Wed, 2005-03-23 at 13:46, Eric Wieling aka ManxPower wrote: Bart Van Daal wrote: Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? That's because you do NOT need a Zaptel timer for Playback, Background, or MoH. No but when I installed my new server I had installed zaptel but left it unconfigured and this is the exact same problem I had. By enabling ztdummy everything started working. I think because zaptel was there it tried using it for timing but as the card was not configured it caused the problem. Asterisk was complaining about not being able to find a timing source. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium support quality: Excellent
On March 23, 2005 07:37 am, Michael George wrote: I was told to try changing PCI slots (I haven't had a chance to do that yet), but since the TDM cannot share IRQs with anything else, changing slots might just put it into a conflict situation. This one could be sticky... As I am learning more and more of the zaptel code I think the *right* solution is to have the driver recognize that it already has a zaptel timing source and turn off the timer on subsequent cards, using the first card detected as the sole generator of interrupts. I've got a few other things on my plate, however, so I haven't been able to really test this. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] who has purchased a V400 card from Varion ?
On March 23, 2005 08:05 am, Ernest Stokes wrote: We are a small shop that had one t100p card, when it came time to expand to a second card we found the price had been raised to $599 from $499 for the single port. The 4 port cards from varion are $699 on special. I believe that I can get it working without any support from varion. Perhaps so, but Varion's got a good deal going -- defer all support to the lists. :-) It would be irresponsible of me to buy a 1 port for the office at that price. Plus I think 1 card instead of 2 would be a better solution in my server. I support Digium any way I can ( t100p, plus 2 TDM cards/x100-non clone when I was first starting out last year) but $100 buys me 3 more ports. I am not bashing your choice, as it was a judgement call. I was just curious as to why you chose to contact the list first instead of the people you bought the card from. (I'm not withholding help or anything... you didn't give any information to start, but secondary to that is the fact that I don't have any experience with the Varion cards. I was merely curious.) -A. PS - it is considered bad ettiquette to CC the author as well as the list, I am already subscribed so I get two copies. Others may disagree and prefer to be CC'd as well but I believe that they're the minority and should include the specific request in their .sig. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe Upgrade !
Dear All : I have a problem with my MeetMe ,, I need to add the latest update of MeetMe Software so that I can add an announce when entering/exiting the Conference Room .. I saw a lot of WebPages are talking about i option. How can I get the latest update ? and how can I recompile my Asterisk to use the new update ?? Thanks ,,, Eng. Mohamed Farid ,, Mediterranean Smart Cards Company ,, Telecommunication Security Administrator ,, * Email : [EMAIL PROTECTED] (Phone : +20 2 3331439/+20 2 3331400 7 Fax : +20 2 7621164 Mobile : +20 0122258350 Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are the addressee, you may not read, copy, use or store this e-mail in any way, or permit others to. If you have received it in error, please contact Mediterranean Smart Cards Company :+202333 1400 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playback of sound files but no sound
Thank you for the answer, I'm using a simple sip-configuration with 3 ip phones. 2 phones are micronet equipment and the last phone is connected to a grandstream handytone-286 box. Using the micronet phones I hear sound and mp3 but it seems the grandstream is having problems with it. I can make calls with the device but hear no soundfiles. thanks again for the quick responses, Bart -Original Message- From: Gareth Blades [mailto:[EMAIL PROTECTED] Sent: woensdag 23 maart 2005 15:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Playback of sound files but no sound On Wed, 2005-03-23 at 13:46, Eric Wieling aka ManxPower wrote: Bart Van Daal wrote: Thanks for your answer, I've compiled and loaded 'ztdummy' but still no sound. here's the relevant portion of lsmod: ztdummy 2464 0 (unused) wcusb 19552 0 (unused) zaptel178752 0 [ztdummy wcusb] i810_audio 28824 0 (autoclean) ac97_codec 16840 0 (autoclean) [i810_audio] soundcore 6436 2 (autoclean) [i810_audio] usb-uhci 25740 0 [ztdummy] Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf) to play back these sound files? That's because you do NOT need a Zaptel timer for Playback, Background, or MoH. No but when I installed my new server I had installed zaptel but left it unconfigured and this is the exact same problem I had. By enabling ztdummy everything started working. I think because zaptel was there it tried using it for timing but as the card was not configured it caused the problem. Asterisk was complaining about not being able to find a timing source. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chanisavail and IAX2
Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten = s,7,ChanIsAvail(IAX2/anton:[EMAIL PROTECTED]) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any tips? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
So far I have tested www.teliax.com, and www.livevoip.com Livevoip has had some problems in the past, and they are fixing their DID problems now. However, their rates are very good. I just signed up for teliax yesterday and so far it has been very straight foward. If you use AMP I have put up a quick howto to get them up and running. Their pay as you go service is a little more expensive than others, but overall it is still a good deal. -Jake On Wed, 23 Mar 2005 19:16:49 +0545, Vicky Shrestha [EMAIL PROTECTED] wrote: Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
You should try Fordvoice http://www.fordvoice.org they are cheaper than broadvoice also. and have the same service. - Original Message - From: Vicky Shrestha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 8:31 AM Subject: [Asterisk-Users] Broadvoice alternatives Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanisavail and IAX2
On March 23, 2005 09:59 am, Anton Krall wrote: But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Just a guess -- is there a qualify statement for that peer in iax.conf? I typically set my qualify to 500 or 1000ms (acceptable lag between me and them, it does NOT determine how often to ping them) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Wed, 23 Mar 2005, Chris Ford wrote: You should try Fordvoice http://www.fordvoice.org they are cheaper than broadvoice also. and have the same service. - Original Message - From: Vicky Shrestha [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 8:31 AM Subject: [Asterisk-Users] Broadvoice alternatives Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-03-23%5Cfcdbdcefe0bd47b985a85fd1f91855feC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls
Dan Levine wrote: Yup that works on our end as well We assign 3 of the same lines to the same phone and it works perfectly. Are you are using the same SIP account for each line on the phone? Could you please post your 7960's SIPmac.cnf file? Asterisk does not support multiple registrations of the same SIP account so I am quite interested in how both of you are accomplishing this. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [0] Wilcard X100P doesn't hang up when in Voicemail() and calling party hangs up.
Rich Adamson [EMAIL PROTECTED] wrote on 2005-03-23 09:08: I have a x100p card and it doesn't detect a hangup from the calling party when going in voicemail(). My PSTN provider is sending open loop disconnect (voltage decrease for a given moment of time). Actually Progress Detection is HIGHLY EXPERIMENTAL so it should not be required to fix this problem. I wonder if disconnect supervision is the same as open loop disconnection. Actually, the service is ok from my POTS provider. I did put an AbsoluteTimeout on the voicemail() but is not a viable solution as for if somebody wants to leave a 10mn message, he wont be able to do so. You might want to look at voicemail.conf and configure the maxsilence for ten seconds or so. That doesn't address your disconnect supervision, but does work. already done that and it doesn't work. I setted it up at 8 seconds just to be sure and even 4... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoicePulse Issues
Adam Robins wrote: So, I switched to IAX2. Now, everything works fine 95% of the time . . . but every once in a while, perhaps 5 seconds into a call or 20 minutes into a call, the call will simply drop. This occurs several times per week with no observable pattern. I have attached an excerpt from the log file at the end of this message. Has anyone else experienced this? Know what is causing it? Has anyone gotten VoicePulse Connect to work with SIP? Hi Admin, I use the connect service from voicepulse ( as I am sure you do, just specifying for future searches ), and I haven't had any of these problems you have mentioned. I do have a problem when the call is connected, there's about half a second of silence about half a second into the call, on every call. I mention it here in case it's related. Honestly, my first instict says this is a firewall problem. Is that at all possible with your setup? Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice alternatives
If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. www.servers-r-us.com Speaking of website being down, I get the following error when trying to check prices on your website: Source data is temporarily unavailable. Please consult this page later on, or take contact with us through the CONTACT page. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 and multiple simultaneous calls
Matthew Boehm wrote: Dan Levine wrote: Yup that works on our end as well We assign 3 of the same lines to the same phone and it works perfectly. Are you are using the same SIP account for each line on the phone? Could you please post your 7960's SIPmac.cnf file? Asterisk does not support multiple registrations of the same SIP account so I am quite interested in how both of you are accomplishing this. -Matthew I don't think the Cisco's actually register twice, they register once per 'account'. These phones simply realize that the account information for both lines is the same and assumes (possibly incorrectly) that it should accept calls to that 'account' on all lines with that account. This disables the phones ability to handle two incoming calls per line button however. My work-around, and I'm sure many others too, was to create a -a and -b 'account' for each 'account' and then do dialplan rollover to make the 7940 accept two calls per line button, or 4 simultaneous incoming calls. The 7960 could accept 12 simultaneous incoming calls this way using a -a through a -f 'account'. -Chris PS: Haven't checked this, but the phone may actually register per 'line' meaning it would register multiple times, but since ALL the details of the register are the same, * just treats it as a re-register and neither * nor the phone know the difference, so both 'work together' to produce this effect. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium support quality: Excellent
-Original Message- I have not yet successfully put 2 TDM cards into a system (though I know others have) and I recenly had a problem where loading the TDM driver and starting * would cause the outgoing message to be played way too fast. Uhhm... We've exclusively used digium cards and the few issues we've had were configuration issues... We've never had an issue w/ a card itself. W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Group channel rotation for outgoing call?
Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to rotate the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:Broadvoice alternatives
Vicky Shrestha Wrote Message: 8 Date: Wed, 23 Mar 2005 19:16:49 +0545 From: Vicky Shrestha [EMAIL PROTECTED] Subject: [Asterisk-Users] Broadvoice alternatives To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal - Try NuFone.net Voipjet.com livevoip.com Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chanisavail and IAX2
Yep, I use qualify also with 1000 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Miércoles, 23 de Marzo de 2005 09:15 a.m. To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Chanisavail and IAX2 On March 23, 2005 09:59 am, Anton Krall wrote: But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Just a guess -- is there a qualify statement for that peer in iax.conf? I typically set my qualify to 500 or 1000ms (acceptable lag between me and them, it does NOT determine how often to ping them) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
Vicky Shrestha wrote: I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. I had some problems here, mainly because I was trying to use g729 and broadvoice will only accept g711. Other than that, configuration itself took about 10~15 minutes with some google search to fix my mistakes... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P and echo
Peter Svensson wrote: On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote: I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the Definity), and cell phones are great. When I call outside of the office to POTS lines (like my home), there is a most noticeable echo of my voice. The party on the line hears no echo. Any efforts on the configuring the SIP softphones, and within zapata.conf, have been for naught. Is this problem common for ISDN-PRI connections? It is a problem you will see when calling an analogue subscriber over a link with a long latency (such as VoIP). The echo will most probably be generated by a 2- to 4-wire hybrid at the far end. In a pure amalogue/tdm path you would perceive the reflected energy as a plesent sidetone. As soon as the latency increases to 50-100ms the refelcted energy will be perceived as an echo instead. The options available to you are to live with the echo of your own voice or to insert an echo canceller at the pstn interface. Asterisk includes an echo canceller that may or may not be good enough. It seems to like some pstn interfaces and not others. If the Asterisk echo canceller is not enough you may consider an expensive inline echo canceller. The definity has echo cancellation. Try turning that on. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: how to sip-h323 using asterisk-oh323-0.7.1
I also have problems the same with you.
I can find my asterisk registered on my GK's status port(7000).
And I make a call from my XPro to SIP and SIP to Asterisk, then
Asterisk calls to a H323 phone via GNUGK.
I can find the CDR message on GK's status monitor. But I the GK only
first ACF(tail with false) and unconnected CDR on my GK.
Do you solve your problem? Can you share me your asterisk config such
as extensions.conf and oh323.conf, gatekeeper.ini for me to refer?
Please.
Best Regards
Charles
On Thu, 10 Mar 2005 23:33:53 -0800 (PST), Kamran Ahmad [EMAIL PROTECTED]
wrote:
hello
i am using my own gnugatekeeper as a gatekeeper for my
asterisk. asterisk is registering successfully with
Gnugatekeeper. but it is not transfering call to
gnugk.
any one guide me who to do this
--
SJPhone(sipSoftPhone using sip)-asterisk
asterisk(conversion from sip - h.323)
asterisk(send h.323)-GnuGK
GnuGk-SoftPhone(h.323 OpenPhone)
-
on GnuGatekeeper side
gatekeeper.ini
[Gatekeeper::Main]
Fourtytwo=42
TimeToLive=600
[RoutedMode]
GKRouted=1
H245Routed=0
CallSignalPort=1721
[RasSrv::PermanentEndpoints]
192.168.0.203=xyz;123
[GkStatus::Auth]
rule=allow
on asterisk
oh323.conf
---
;
; Configuration file of OpenH323 channel driver
;
;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
listenAddress=192.168.0.203
listenPort=1719
connectPort=1719
tcpStart=1
tcpEnd=2
udpStart=1
udpEnd=2
fastStart=yes
h245Tunnelling=no
h245inSetup=no
inBandDTMF=yes
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
tos=lowdelay
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=1
libTraceFile=stdout
gatekeeper=192.168.0.153
gatekeeperPassword=test1
accountcode=test1
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
context=default
[xyz]
type=h323
prefix=123
context=default
alias=1234
context=default
;-
; Specify and configure CODEC related
; options
;-
[codecs]
codec=G711U
frames=20
extensions.conf
--
[default]
exten=2000,1,Dial(SIP/${EXTEN})
exten=3000,1,Dial(SIP/${EXTEN})
exten=_123,1,Dial(SIP/${EXTEN})
exten=_321,1,Dial(OH323:h323/[EMAIL PROTECTED]:1719|30|r)
sip.conf
--
[2000]
host=dynamic
type=friend
dtmfmode=INFO
canreinvite=no
[3000]
host=dynamic
type=friend
dtmfmode=INFO
canreinvite=no
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Best Regards
Charles
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RE: [Asterisk-Users] Group channel rotation for outgoing call?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro G Sent: Wednesday, March 23, 2005 10:43 AM To: Asterisk Subject: [Asterisk-Users] Group channel rotation for outgoing call? Hi, If I have a PRI with all channels grouped in group=1, I understand when I want to make an outgoing call that asterisk takes the first channel available. Is there any possiblity to rotate the channel taken? I was searching in Wiki but I could not find nothing about. Thanks, Alejandro --- From the Wiki. Link: http://www.voip-info.org/wiki-Asterisk+zap+channels Dialing a Group In the Zap Channel Module's configuration file (zapata.conf), you can define groups of Zap channels that get treated as a single channel as far as the Dial command is concerned. You specify which of four methods the Zap channel module is to use to select a non-busy channel from the channel group by prefixing the group number with one of the letters g, G, r, or R: g: select the lowest-numbered non-busy Zap channel (aka. ascending sequential hunt group). G: select the highest-numbered non-busy Zap channel (aka. descending sequential hunt group). r: use a round-robin search, starting at the next highest channel than last time (aka. ascending rotary hunt group). R: use a round-robin search, starting at the next lowest channel than last time (aka. descending rotary hunt group). The round-robin searches make the Zap channel module start looking for an available channel from a different channel number each time. For each channel group, the Zap channel module keeps track of the last round-robin start point, and this time starts checking availability from either the next (lowercase r)) or the previous uppercase R channel in the group. Which channel it actually finds available (if any) does not affect the starting point for the next round-robin search. Calls to the Dial command using ordinary (g or G) group selections do not affect future round-robin starting points either. For example, if you have defined channel group 2 as containing Zap channels 1, 2, 5 and 8, and the last round-robin search for this group (group 2) began searching from channel 5, this is the order of searching that the Zap channel module will use for the four possible selection methods: Dial(Zap/g2...): Looks in order 1, 2, 5, 8 Dial(Zap/G2...): Looks in order 8, 5, 2, 1 Dial(Zap/r2...): Looks in order 8, 1, 2, 5 Dial(Zap/R2...): Looks in order 2, 1, 8, 5 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Group channel rotation for outgoing call?
Yes there is. Try:
Dial(Zap/G1/w${EXTEN})
The capital G makes * grab channels in the opposite
order as little g. Hope that helps.
Richard
--- Alejandro G [EMAIL PROTECTED] wrote:
Hi,
If I have a PRI with all channels grouped in
group=1, I understand when I
want to make an outgoing call that asterisk takes
the first channel
available.
Is there any possiblity to rotate the channel
taken? I was searching in
Wiki but I could not find nothing about.
Thanks,
Alejandro
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RE: [Asterisk-Users] Setting MWI on legacy PBX
- Original Message - From: Brian S. Adelson [EMAIL PROTECTED] You could probably utilize vmnotify to do exactly what you are looking for: http://mikecathey.com/code/vmnotify/ Thanks. I may use that as a starting point if my home-grown solution doesn't work. I have a shell script that seems to be doing an okay job so far, though I won't know for sure until the system is in full use. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] speex 1.1.7 crashes asterisk 1.0.6
Hello, I've installed speex 1.1.7 and asterisk 1.0.6 from Gentoo's Portage and I'm experiencing asterisk crashes whenever I try to make a connection from my X-Lite client under wine to asterisk using the speex codec. I know speex is being attempted because SPX lights up on the X-Lite display. Also, I know that speex is causing the crash because X-Lite works fine if I use GSM, ILBC, or ULAW. Speex seems to be installed in asterisk too, because I can run SHOW TRANSALTION and get speex output. Has anyone had this happen before? Has anyone successfully used speex 1.1.7 with asterisk 1.0.6? Do I need to try another speex revision? Thanks! -- Jesse Guardiani, Systems Administrator WingNET Internet Services, P.O. Box 2605 // Cleveland, TN 37320-2605 423-559-LINK (v) 423-559-5145 (f) http://www.wingnet.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where to put the modules to start on boot?
Sorry for this kinda n00b question but I've been looking through the wikis but didn't find the answer. All info pages tell you how to load modules from the commandline but what is the `proper' way to do this at boot time? My gentoo box has a /etc/modules.autoload.d/kernel-2.6 but there is no such thing on RedHat boxes. Where do you put the module load and init commands on a RHEL 4 box and where to put it on a RHEL 3 box? Thanks! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not server divas) with asterisk. First thing to do is to load the module. If you have two of these cards, you should do it like that: modprobe -v hisax protocol=2,2 type=11,11 And now you can have up to 4 incoming calls with two cards (try calling yourself and see if anything gets into your syslog - you should have ignored calls even if asterisk isn't running). Then configure your asterisk to use i4l (don't use chan_capi) - do it in modem.conf: (...) driver=i4l (...) msn=your_msn_number and that's it (you still need to configure your ISDN devices to allow incoming calls, for example, using conf-isdn-account - don't forget to set SECURE=off etc. ISDN settings). Tomek ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Settings to improve voice quality?
Im using Broadvoice and just got it working last night. Once noticable annoyance is that the audio quality is pretty poor. There are pops and volume fading. Are there settings that will improve this? JD ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chanisavail and IAX2
This is really weird.. Ive tried all combination for doing ChanisAvail(IAX2/) with no luck.. * still thinks the channels is not available eventhough I can dialout using it. Any pointers? Anybody using this config? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Miércoles, 23 de Marzo de 2005 09:46 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Chanisavail and IAX2 Yep, I use qualify also with 1000 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Miércoles, 23 de Marzo de 2005 09:15 a.m. To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Chanisavail and IAX2 On March 23, 2005 09:59 am, Anton Krall wrote: But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Just a guess -- is there a qualify statement for that peer in iax.conf? I typically set my qualify to 500 or 1000ms (acceptable lag between me and them, it does NOT determine how often to ping them) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to put the modules to start on boot?
On Wed, 2005-03-23 at 17:15 +0100, Remco Barende wrote: Sorry for this kinda n00b question but I've been looking through the wikis but didn't find the answer. All info pages tell you how to load modules from the commandline but what is the `proper' way to do this at boot time? My gentoo box has a /etc/modules.autoload.d/kernel-2.6 but there is no such thing on RedHat boxes. Where do you put the module load and init commands on a RHEL 4 box and where to put it on a RHEL 3 box? See, this isn't an asterisk specific questions. It becomes a distro specific question. Continue looking for a /etc/modules.conf or /etc/modules or even /etc/conf.modules -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why even have set CallerID option?
Why even have the ability to set callerid name/number if end offices don't honor it? For example, I have a SIP UA registered and in the sip.conf I have: callerid=Mark Mane 2815692712 When that phone makes an outbound local call, asterisk will terminate it on PRI connected to asterisk box to Time Warner. When the called party looks at their caller id display screen it shows the number that is in sip.conf, but does not show the name I have set in the sip.conf; instead it shows our company name (since we own the number). If it is the responsibility of the last end office to do a data-dip and select out the name, then that means I cannot control the callerid name, correct? So I guess that callerid name is only useful for VoIP-VoIP calls that go thru asterisk? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to put the modules to start on boot?
Steven Critchfield wrote: On Wed, 2005-03-23 at 17:15 +0100, Remco Barende wrote: Sorry for this kinda n00b question but I've been looking through the wikis but didn't find the answer. All info pages tell you how to load modules from the commandline but what is the `proper' way to do this at boot time? My gentoo box has a /etc/modules.autoload.d/kernel-2.6 but there is no such thing on RedHat boxes. Where do you put the module load and init commands on a RHEL 4 box and where to put it on a RHEL 3 box? See, this isn't an asterisk specific questions. It becomes a distro specific question. Continue looking for a /etc/modules.conf or /etc/modules or even /etc/conf.modules RH has /etc/modprobe.conf on RHEL4. The modules themselves of course go in /lib/modules but the commands to laod them go in /etc/modprobe.conf (they changed from /etc/modules.conf with the change to 2.6). Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why even have set CallerID option?
Matthew Boehm wrote: Why even have the ability to set callerid name/number if end offices don't honor it? For example, I have a SIP UA registered and in the sip.conf I have: callerid=Mark Mane 2815692712 When that phone makes an outbound local call, asterisk will terminate it on PRI connected to asterisk box to Time Warner. When the called party looks at their caller id display screen it shows the number that is in sip.conf, but does not show the name I have set in the sip.conf; instead it shows our company name (since we own the number). If it is the responsibility of the last end office to do a data-dip and select out the name, then that means I cannot control the callerid name, correct? Close enough, yeah. So I guess that callerid name is only useful for VoIP-VoIP calls that go thru asterisk? -Matthew Yup. Which is actually very helpful for me. My offices are going to have about 50 phones, and the callerid on the phones will be extremely helpful for sip-sip calls. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need some help
Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. Thanks for any advice Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Thanks Yves, Thanks for this good news, that digium going to start h323 channel soon. Oh this is at least one hope i can see. Bashir - Original Message - From: Yves [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 22, 2005 11:57 PM Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable If you use open-source software, you have to accept that sometimes project need some times to be stable and have all features. OH323 works - even if there are still a few bugs - and the people around the project are working hard to make to work even better. If you want something that work now, with support , there are plenty of commercial products. I suggest you continue trying oh323, or be ready to pay. I don't know the existence of any other open-source that can do this. Except a post on the dev-mailing telling that Digium was coding his own h323 channel module. WaitSee. Yves Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need some help
Hello, what is the benefit of your scenario #2? I'm not understanding what it adds for you... -yair On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED] wrote: Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. Thanks for any advice Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need some help
Duh, i'm an idiot. I meant scenario #1. -yair On Wed, 23 Mar 2005 18:52:28 +0200, Yair Hakak [EMAIL PROTECTED] wrote: Hello, what is the benefit of your scenario #2? I'm not understanding what it adds for you... -yair On Wed, 23 Mar 2005 08:49:37 -0800 (PST), Alex [EMAIL PROTECTED] wrote: Hi all I have a couple of questions maybe you guys can help me with them I have sip phones , SER server , Asterisk. what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. Thanks for any advice Do you Yahoo!? Yahoo! Small Business - Try our new resources site! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Wed, 2005-03-23 at 17:18 +0100, Tomasz Chmielewski wrote: I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not server divas) with asterisk. Last time I tried - there were a few problems... 1 - Outbound DTMF - never made it... ie You can not interact with someone else's IVR (DTMF controlled systems) 2 - Inbound DTMF - Certain voices would be interpreted as DTMF - which is fine until they sounded like a '#' - and got transfered (some strange reason - my wife's voice - especially when she got angry) I believe that there was some sort of patch for (2) but never heard of a fix for (1) Has this changed at all??? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable
Hi George I did install and checkup several times, but some times h323 gateway or softswitch cant accept my call and was able to accept call but no sound. so can you help me please to implement a h323 solution. You may contact with me if you want. Thanks Bashir Call. 1-604 323 7991 Mail. [EMAIL PROTECTED] - Original Message - From: George K. Konstantoulakis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 3:11 AM Subject: Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable Hello Bashir, what kind of problems are you having with oh323 ? George Bashir Ullah - www.Lamsre.Com wrote: Hi All * lover. This is not a question only this is a request to all SIP and Asterisk user . I am also with asterisk last few month and providing callingcard soluation. most of the SIP or IAX provider asking very high price which is really tough to resell in market. but still there is some h323 provider offering good price. so as a asterisk user i tried so many times and now give up to implement oh323, h323 by asterisk. i am sorry and also there is very may be none user for asterisk with h323. Thats why need a seperate soluation and open source for converter h323 to sip vies-versa for asterisk user. Is it possible in near future. or is there any solution already done with is open source. Thanks for your time to read this mail. Bashir ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming response and external access
What are the procedures required to a. Call my Asterisk box from eyeBeam/Ineen b. Connect to my Asterisk box as a proxy from eyeBeam/Ineen I setup IPCop to only allow my home (static) IP as a security measure. I can call extension on the LAN fine, and call external numbers through the modem, it received phone calls from the phone line no problem, just the External Internet to Asterisk that the problem. Not to state the obvious, but I assume you've followed http://www.voip-info.org/tiki-index.php? page=Asterisk%20firewall%20rules to ensure all the negotiation and media ports are open and forwarded to *, correct? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
[EMAIL PROTECTED] (Bruce Komito) writes: If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. And he could also lose that flash animation when promoting to an opens-source/linux audience. The fordvoice web site has a big blank blotch where I assume some information presented in flash format would go. Not exactly effective marketing... -wolfgang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GR-303 from Central Office supported?
Rich Adamson wrote: I'm a little confused on whether the GR303 support in * will accept calls from a Siemens central office that has GR303. I don't know for sure (sorry for responding anyway), but I believe that Asterisk's GR-303 support is the 'network' end only, so that it can control access concentrators. That would mean that is does not have the ability to act as an access concentrator. Also keep in mind that GR-303 encompasses far more than just call control; there are also provisioning and OAM parts of the protocol, and a switch tech that sees a GR-303 trunk heading out of their switch may very well assume that the gear at the other end is theirs and they can re-provision it :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice alternatives
If you never had the same problem at one time or another Please Stand Up... Thanks for the Info thoe... Chris Ford - Original Message - From: Nabeel Jafferali [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 10:30 AM Subject: RE: [Asterisk-Users] Broadvoice alternatives If you're going to promote your product, you might consider making sure your web site is up, before giving out the URL. www.servers-r-us.com Speaking of website being down, I get the following error when trying to check prices on your website: Source data is temporarily unavailable. Please consult this page later on, or take contact with us through the CONTACT page. Nabeel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Need some help
what is the best way to do that (also with accounting and authentication). which one of those options 1) sipphone - SER - ASTERISK - SER - PSTN 2) sipphone - SER -ASTERISK -PSTN on the first option i am trying to return the call to the ser after it's pass the asterisk for some routing solutions and accounting. but i have some problems to hear the other side. I really don' t see the point of going through SER before PSTN SER is mostly good for offloading REGISTERs. Apart from that, asterisk can handle the rest, so I'd forget about 1) roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP messagse
hi; say i have two users A and B registered with asterisk. A sends an INVITE to B thru *. My question is how can i re-write some of the parameters in the SIP or SDP message sent from A to B? thanks m.smadi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why even have set CallerID option?
[EMAIL PROTECTED] (Matthew Boehm) writes: Why even have the ability to set callerid name/number if end offices don't honor it? VOIP is bigger than just PSTN-gatewayed calls via some specific company. The end goal is to connect the VOIP islands directly. That is already happening at some large companies where they call their supplier directly on a purely voip link. For a concrete example look at the sip-edu program. It is a growing group of universities that exchange SIP calls directly. (Some even have their asterisk and SER config notes on line.) In all cases the caller's calling-number and calling-name stuff will get passed to the callee. http://voip.internet2.edu/SIP.edu/ -wolfgang ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC: perl / mysql or me???
I did not get any hint to my first try, ... can somebody help me?
I try to change something in ASTCC, but I am now totally blind,
I hang on one line now. I changed:
vpbx:/var/lib/asterisk/agi-bin # diff astcc-original.agi astcc.agi
22c22
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN})
---
# exten =
_00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},${TARIFF},${EXTEN})
35c35
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},BALANCE,1)
---
# exten = _00X,1,DeadAGI(astcc.agi,${CALLERIDNUM},BALANCE,'',1)
273,274c273,276
I added one parameter ${TARIFF}
my ($number) = @_;
my $sth = $dbh-prepare(SELECT * FROM routes WHERE .
$dbh-quote($number) . RLIKE pattern ORDER BY LENGTH(pattern) DESC);
---
my ($number, $tariff1) = @_;
my $sth = $dbh-prepare(SELECT * FROM . $tariff1 . WHERE
. $dbh-quote($number) . RLIKE pattern ORDER BY LENGTH(pattern) DESC);
print STDERR sth = $sth\n;
277a280
print STDERR res = $res\n;
413c416
($calleridnum, $phoneno, $quiet) = @ARGV;
---
($calleridnum, $phoneno, $tariff, $quiet) = @ARGV;
521c524
print STDERR Phone number is $phoneno\n;
---
print STDERR 1. Phone number is $phoneno\nTariff is
$tariff\n;
526c529
$numdata = getphone($phoneno);
---
$numdata = getphone($phoneno, $tariff);
554c557,560
$numdata = getphone($phoneno);
---
print STDERR 2. Phone number is $phoneno\nTariff is $tariff\n;
$numdata = getphone($phoneno, $tariff);
print STDERR 2.a numdata = $numdata\n;
print STDERR 2.b Matching pattern is $numdata-{pattern}\n;
555a562
print STDERR 2.c numdata = $numdata\n;
556a564
print STDERR 2.d quiet = $quiet\n;
vpbx:/var/lib/asterisk/agi-bin #
What happens is, when I use the $TARIFF=routes (what was the original
name) it works! If I use the new table name I had added to the database,
than it does not work!
The database has both tables routes and newrates.
With routes I get: You have so much money, your call cost
With newrates I get: You have so much money left, I am sorry that is
not a recognized number
I created the newrates table via mysqldump, changed table name
everywhere and changed the rate, inserted the new table with mysql, ...
I tried to reload mysql, ...
Please, enlighten me!!!
bye
Ronald
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[Asterisk-Users] Random use of Sip peers
I know that I can use g or G for Zap lines, but how can I use group and more exactly random lines of a group? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanisavail and IAX2
it doesn't work with current CVS, it works with 1.0.7 - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, March 23, 2005 9:59 AM Subject: [Asterisk-Users] Chanisavail and IAX2 Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten = s,7,ChanIsAvail(IAX2/anton:[EMAIL PROTECTED]) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any tips? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
