[Asterisk-Users] Compile Problem
I am trying to compile the asterisk and if fails at the end on: make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx'gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread`/usr/lib/gcc-lib/i486-slackware-linux/3.3.2/../../../../i486-slackware-linux/bin/ld: cannot find -lXextcollect2: ld returned 1 exit statusmake[1]: *** [pbx_gtkconsole.so] Error 1make[1]: Leaving directory `/usr/src/asterisk-0.5.0/pbx'make: *** [subdirs] Error 1 Anyone know what is wrong? Linpri and zaptel compiled just fine. This is linux slackware 2.4.23 all the latest from 9.1 slackware distrib fresh system install. Thanks, John
Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download
Hi all, Please send all the feedback related to the ActiveX version of DIAX directly to me, not to the list. I'll try to handle each request individually. Thank you for your understanding, Dan P.S. The first version is tested and must works on Windows 98SE/2000/XP.
[Asterisk-Users] asterisk with a third party gateway
Hello Can asterisk be configured as a PBX with a third party gateway (cisco router 3640 running Cisco call manager express). The cisco gateway will only interface the PSTN and asterisk, so the cisco routerwill handle incoming and outgoing calls. I would like to do this as we have the hardware like VIC ISDN cards with us. The main reason for going with asterisk is to hook up with a VOIP service provider. Any help in this regard will be greatly appreciated Thanks Deepak
[Asterisk-Users] ActiveX DIAX demo available online
Hi, For the ones who does not have a web server to test, there is a demo of DIAX ActiveX available online (for another 4 hours) at: http://193.231.214.47:25380/dax.htm If you accept to enable unsigned ActiveX to be downloaded and run on your PC, then you can play with it (on your own risk..:-)) You will be propted at first run for the credentials. You can use your IAXTEL account to play with it. All personal data (user and pass) will be keept in a file on your system, not on the server. Further, if yo start taht page you will not be asked for the credentials. If you want to change them, click on the 'C' box in the upper right corner. If you have tried before the ActiveX locally, please delete any refference to it from the registry (HKLM) before testing this demo. Enjoy it, Dan P.S. I am not responsable for any damage that can be done on your system using this ActiveX (diax.ocx). It has been tested, but has no guarantee. Please DO NOT USE IT on a production system. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Hi all, Clipcomm - looks interesting, you get NAT/Ethernet/Analog line (to SIP) D-link DVG-1120M/H/S - this is also on the lines of what I'm looking for, lets you connect standard analog phone directly to it, has NAT and an Ethernet port - anyone ever tried this with * ? What I am exactly trying to find is an easy to setup/'elegent'/cheap (I know I might be asking for too much) solution to provide SIP connectivity (while maintaining Internet connectivity from PC) considering two scenarios: 1. Cable ISP user with single IP over DHCP (the user already has a network card in his PC): this is quite easy, the D-link DVG-1120 will do the trick - PC goes to to the Ethernet plug, analogue phone goes directly to the RJ-11 plug (nice) 2. ADSL (PPPoA) user with a USB connected modem, single IP over DHCP (no network card in his PC): this is more tricky - Intertex IX66+PF may do the trick although is quite expensive, I've also found SMC Barricade ADSL Modem Router with similar features at a lesser price. This still calls for a ATA286 analog phone adaptor if the guy is to use his normal telephone. The ideal case would be a a small cheab box with an Ethernet/ADSL WAN plugs on one end and an Ethernet/USB/Analog (from SIP) plugs on the other end ;-) Thanks, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Philipp von Klitzing Sent: Friday, December 19, 2003 1:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem) Hi! I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for the telephone. Something that would combine the functionality of a (adsl modem+) router and a SIP telephone adaptor in one box. This is not exactly what you are asking for, but its getting close: http://www.voip-info.org/tiki-index.php?page=VOIP+Phones Look at the Clipcomm devices (Korea). I'd be interested in any reports experiences. Next to that the newer Grandstream firmware now supports PPPoE (but now routing or port forwarding). Not sure what the CISCO devices offer in this respect. I wouldn't want to have ADSL modem router coupled in one device, that'll make your router useless if you move to cable modem etc. So better not integrate too many task into one piece of hardware. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with a third party gateway
On Mon, 2003-12-22 at 06:50, Deepakumar JV wrote: Hello Can asterisk be configured as a PBX with a third party gateway (cisco router 3640 running Cisco call manager express). The cisco gateway will only interface the PSTN and asterisk, so the cisco router will handle incoming and outgoing calls. I would like to do this as we have the hardware like VIC ISDN cards with us. Yes you can. Even you dont need CCM just sip-ua and the correct dialpeers. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fedora core 1 install problem
Hi Ernest, I have installed as you described, and now it worked. Seems that installing a minimum system and afterwards installing the necesary packages with their dependencies seems to not have worked for me Thanks for the help all... David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Ernest W. Lessenger Verzonden: woensdag 17 december 2003 18:54 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] fedora core 1 install problem At 09:20 AM 12/17/2003, you wrote: Hi, I am trying ti install an asterisk system on fedora core 1. During the make of asterisk I got the folowing problem: bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe Does anybody know how to solve this? David I can't, but I can tell you that I installed Fedora Core 1 with Development Tools, Editors and Kernel Development, and that asterisk installed without any problems at all. --Ernest \/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o tdd.o tdd.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o acl.o acl.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o rtp.o rtp.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o manager.o manager.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-12/17/03-17:46:24\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP-c -o asterisk.o asterisk.c bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c make: *** [ast_expr.c] Broken pipe [EMAIL PROTECTED] asterisk]# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicetronics
Hi list, Is there anyone using the voicetronics openline4 with asterisk. Does this card work ok for 4 port analoge fxo? Thanks, Tjapko. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call files
I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called bad! What I would like to get round this is probably the reverse I dont want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? It is probably something very simple I am missing! Nick
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
I'm missing something here. I've put the following in extensions.conf and a
few variations thereof. I've taken the sample configs and added to them, so
when I dial 2200 from netmeeting * answers and runs me through the demo
announcements.
The pots extensions 2200 and 2107 (TDM400) work fine calling each other and
cause netmeeting to ring when I dial 3100, but the audio is one way
pots-netmeeting when I answer in netmeeting.
If it is an RTFM situation please give me a URL, pretty postcard to anyone
than can help me.
extensions.conf
[incoming-h323]
exten = 3001,1,Dial,OH323/192.153.153.64
exten = 3001,2,Busy
exten = 3001,102,Busy
[default]
include = incoming-h323
include = demo
exten = 2107,1,Dial(Zap/32,20)
exten = 2107,2,Voicemail(u2107)
exten = 2107,102,Voicemail(b2107)
exten = 2200,1,Dial(Zap/33,20)
exten = 2200,2,Voicemail(u2200)
exten = 2200,102,Voicemail(b2200)
At 13:42 19/12/03, you wrote:
bam wrote:
I've read through the archives and have picked up that * does not need a
gatekeeper to talk directly with an H323 handset to send and receive calls.
I'm trying to go PSTN*-H323 and all the examples that I can find
use a gatekeeper. Are there any examples or hints for doing it without
the gatekeeper?
many thanks in advance
Brian
[your_context]
exten = _9XX,1,Dial,H323/78632${EXTEN:[EMAIL PROTECTED]|30
exten = _9XX,2,Busy
exten = _9XX,102,Busy
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Best Regards,
Pavel Litvinenko.
ICQ: 16224754
Ph: (8632) 923962, 923640
sip:[EMAIL PROTECTED]
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[Asterisk-Users] windows messenger and DTMF
Hello All, Another question todayJ I have just started playing with windows messenger and asterisk following the little how-to from the asterisk web-site work well good sound quality but you cannot put people on hold transfer them or send DTMF (to get asterisk to do the transfer) or am I missing something this makes quite a poor soft phone if this is the case? Thanks Nick
RE: [Asterisk-Users] call files
Hi, Why don't you turn the process around: 1/ file is created 2/ internal number is called 3/ internal party answers 4/ internal party hears ringing as the external party is being called. Ofcourse everything depends on how you have built your dialplan since you'd need to have access to an context that can do outbound dialling by number. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called - bad! What I would like to get round this is probably the reverse - I don't want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] windows messenger and DTMF
There are 2 ways to place a call in MSN messenger, either place a voice message or place a call. Place a voice message is like chat, but you put in your [EMAIL PROTECTED]but if you use this option, you can't send dtmf digits, place a call drops down a dtmf keypad. If you don't get the keypad, then make sure your MSN registry option CorpPC2Phone is set to 1. Also, unless they fixed it, MSN messenger can't be put on hold (or you can, but you can't get it back again!) Lee Goodman - Original Message - From: Nick Knight To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 9:26 AM Subject: [Asterisk-Users] windows messenger and DTMF Hello All, Another question todayJ I have just started playing with windows messenger and asterisk following the little how-to from the asterisk web-site work well good sound quality but you cannot put people on hold transfer them or send DTMF (to get asterisk to do the transfer) or am I missing something this makes quite a poor soft phone if this is the case? Thanks Nick
[Asterisk-Users] no monthly fee
Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no monthly fee
http://www.iconnecthere.com and http://connect.voicepulse.com as long as you don't need an incoming phone number. - Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 10:34 AM Subject: [Asterisk-Users] no monthly fee Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no monthly fee
Yes, Check out VoicePulse, they bill by the minute with no monthly fee. IConnectHere also allows plans that bill by the minute, however there is a 0.65/month fee. Which, considering the ammount you'll save, is very very tiny. Regards, Brent On Mon, 22 Dec 2003, Hector Q.-datafull wrote: Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audio format for announcements
Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec? 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? Thanks, Sean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no monthly fee
www.nufone.net On Mon, 22 Dec 2003, Jim Flagg wrote: http://www.iconnecthere.com and http://connect.voicepulse.com as long as you don't need an incoming phone number. - Original Message - From: Hector Q.-datafull [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 10:34 AM Subject: [Asterisk-Users] no monthly fee Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no monthly fee
i think nufone and xvoip are based on a per min basis prepaid perhaps but no monthly fee there is probably others as well On Mon, 2003-12-22 at 10:34, Hector Q.-datafull wrote: Hi, anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls? thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZTMonitor - /dev/dsp problem
try ztmonitor 1 -v Martin On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicetronics
iTS [EMAIL PROTECTED] wrote: Hi list, Is there anyone using the voicetronics openline4 with asterisk. Does this card work ok for 4 port analoge fxo? Thanks, Tjapko. We are using openline4 for testing purpose, not in production. It seems to work. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SJphone, Asterisk and DTMF tones ...
You are lucky. I'm getting this: -- Incorrect password '1334' for user When I enter 1234. I'm using dtmfmode=rfc2833 and a GS Budgtone 100 phone. Why do I getr 4x while you get 2x ?? use dtmfmode=info (both in sip.conf and in your GS settings, of course) search the archives to find this mentioned often. ;-( Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio format for announcements
- Original Message - From: Sean Adams [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 10:50 AM Subject: [Asterisk-Users] Audio format for announcements Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec? 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? Can't help with the first question, but ulaw/alaw ~= g711, which I think is the biggest eater of bandwidth that I've heard of. See: http://www.voip-info.org/wiki-Codecs - Andrew Thompson http://aktzero.com/ Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue App
Just a note to Mark and others. In queue.conf, there is a reference to announce-markq that I believe comes default uncommented. There is no sample file in /var/lib/asterisk/sounds/announce-markq If there is no file there and/or you misspell the filename and the system can't find the announce file it will hang up on the person calling in. Generates a console log of Channel hanging up on call and caller is going to be pissed. LOL It might be better to do a if/then statement for the announce file so if it gets moved/deleted it wouldn't cause a problem with the system. Just my 2 cents worth. Tim. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToIP (TDD over IP)
Hi! I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. Excuse my ignorance: What exactly is TDD? Is it US specific? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip show peers - disappearing
We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in sip show peers with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P connected to Cisco
You need to have HDLC generic support compiled into your kernel ... I think it's not good to have it compiled in modules ... just embedded in kernel. Martin On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi All, I wish to connect * to a Cisco using a E100P board. When I load the driver I got this error message: -bash-2.05b# modprobe wct1xxp ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows Cisco configuration: isdn switch-type primary-qsig isdn voice-call-failure 0 controller E1 2 framing NO-CRC4 clock source line primary pri-group timeslots 1-31 interface Serial2:15 no ip address isdn switch-type primary-qsig isdn overlap-receiving T302 2000 isdn incoming-voice modem isdn T310 4 isdn send-alerting no cdp enable voice-port 2:D cptone BR I configured my /etc/zapata.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 Any clue? Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicetronics
Thanks, any special configuration requirement? Tjapko. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza Sent: Lunes, 22 de Diciembre de 2003 05:01 p.m. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicetronics iTS [EMAIL PROTECTED] wrote: Hi list, Is there anyone using the voicetronics openline4 with asterisk. Does this card work ok for 4 port analoge fxo? Thanks, Tjapko. We are using openline4 for testing purpose, not in production. It seems to work. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.551 / Virus Database: 343 - Release Date: 11/12/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ToIP (TDD over IP)
Telecommunications Device for the Deaf -Original Message- From: Philipp von Klitzing [mailto:[EMAIL PROTECTED] Sent: Monday, December 22, 2003 11:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ToIP (TDD over IP) Hi! I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. Excuse my ignorance: What exactly is TDD? Is it US specific? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Audio format for announcements
On Mon, 2003-12-22 at 09:50, Sean Adams wrote: Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable. Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection. 1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec? If it is recorded well, and is played on decent interfaces, it won't sound bad. If you so wish, you can store these as wav files in ulaw or alaw format. Look at the codec list and decide what you wish to do. I'm sorry that I have forgotten the name of the person I helped before, but I hosted a prompt for a person so they could see how much of a difference a digital link makes for sound quality. I don't think you would notice a problem no matter what codec within reason if you have a good link. 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? All phones should support ulaw or alaw. Those are pretty much the least compressed you will get. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFAX application
Hi mack_jpn I think problem is CFR 84 sending. In console appears that CFR 84 si sent but te other fax doesn't receive CRF 84, and then RXFAX is waiting for the fax but the other fax doesn't send it ever. I try to see source code. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Masakazu Nakano Enviado el: domingo, 21 de diciembre de 2003 5:37 Para: [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] RxFAX application Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 Sergio Serrano Revuelto [EMAIL PROTECTED] wrote: Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: snip 5 (0.01679,-0.16590) - 0.02781 6 ( -0.04451, 0.75304) - 0.56904 7 ( -0.01415,-0.29305) - 0.08608 Fast carrier down Segmentation fault And i obtain 8 byte tif file. Any Idea? I have installed tiff-3.5.7 and spandsp-20031021. I get same result. but the end part looks like that. Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down -- Hungup 'Zap/1-1' with no segfault I'm tryed with tiff-v3.6.0 ( use with tar balled headers ) and spandsp-20031021 Does anyone have good result? Regards. mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ToIP (TDD over IP)
TDD is a very simple teletype like unit for Telecommunications for the Deaf Which is hooked up to a telephone line with an acousic coupler remember these ? It transmits with 45 baud / BAUDOT code , but unlike regular modems the carrier is removed once the key has been released. TDD is supported by most goverment agencies, emergency services (911) etc. here in the US. Alfred. Hi! I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. Excuse my ignorance: What exactly is TDD? Is it US specific? Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN-PRI - WCT1XXP error
Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf: span=1,0,0,ccs,hdb3 nethdlc=1-15 loadzone = us Thanks in advance, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
The registry expires after sime time. You can set the default expirey and max in sip.conf. It's up to your phone/sip device to reregister after the registration expires. Martin On Mon, 22 Dec 2003, Jonathan Tew wrote: We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in sip show peers with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audio format for announcements
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Adams Sent: Monday, December 22, 2003 10:50 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio format for announcements [...] 2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls? Use g.711ULAW. I belive it is about an 87k uncompressed stream. Sounds better than toll quality to me. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToIP (TDD over IP)
On Monday 22 December 2003 10:12, Philipp von Klitzing wrote: Hi! I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. Excuse my ignorance: What exactly is TDD? Is it US specific? It's a specification for sending words over a normal telephone, normally used by the deaf. It resembles the old-style modems in that the handset is interfaced with a microphone and speaker. This allows TDD to be used with payphones, which do not have an RJ-11 interface. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Audio
My Strandstream BT100 is working OK for both inbound and outbound now except that when you speak into the handset you cannot hear your own voice in the earpeice. It works OK, the other end can hear the call but most telephone users have become used to hearing their own voice. Is this something I can fix or is it a feature of the GS phone? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting audio gain for SIP extensions?
Is there a way to set to audio gain for each SIP extension? I see in the docs this can be done for zaptel but I don't see it documented for SIP. It would be nice to be able to make the various kinds of extensions have equal volume. = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip show peers - disappearing
My guess would be that the NAT firewall times out and closes the port. Reopening it from the inside is no problem, but access from the outside gets blocked. In order to keep the path open both ways, the client needs to send some kind of messages with the proper IP/port in regular intervals. Alfred. We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in sip show peers with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
Their firewall may be timeing them out. Try adding qualify=60 to each of the entries in sip.conf On Mon, 2003-12-22 at 10:26, Jonathan Tew wrote: We have people connecting to an asterisk box over the internet. They're using the x-lite client behind linksys firewalls. The X-Lite client discovers the firewall no problem and connects to Asterisk without a problem. After connecting the agent shows up properly in sip show peers with the IP address of their firewall, etc. They can receive calls no problem. After some time goes by... they don't show as registered with * any more in the sip show peers. They can still make outbound calls, but can not receive the inbound ones. Anyone have any ideas on this one? Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZTMonitor - /dev/dsp problem
Well thats broken.. we have a bug on bugs.digium.com over this. bkw On Mon, 22 Dec 2003, Martin Pycko wrote: try ztmonitor 1 -v Martin On Sat, 20 Dec 2003, Daniel Bichara wrote: Hi, I am trying to run ZTMonitor to get debug info from my E100P board but I got the following message: -bash-2.05b# ./ztmonitor 1 Unable to open /dev/dsp: No such file or directory Cannot open audio ... -bash-2.05b# Thanks, Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call files
I am after using a web crm system which has a button to then get asterisk to dial the contact. For this I was looking at call files, which appear good for the job, I have one small problem with them though. 1/ file is created 2/ external number is called 3/ the external party answers 4/ the external party now hears ringing as you extension is now being called bad! What I would like to get round this is probably the reverse I dont want the people I am calling to hear ringing. For example as soon as it has dialled the receiving end call me, or call me first then call the other extension? It is probably something very simple I am missing! Swap the numbers around. Hello again, I cannot figure this out - just swap them round? I have Channel: CAPI/isdn number::number MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: 101 Priority: 1 But if I swap it round Channel: SIP/User MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: internal Extension: ??? Priority: 1 I have tried various things on the extension but cannot figure out how that would works up to that point as it calls my extension but then obviously fails when it cannot figure out what to do on the return??? Thanks again Nick
[Asterisk-Users] Festival sounds like a steam engine
I tried running the festival app today with little success. I have a working festival installation that does TTS to the linux sound output perfectly. With * it just produces a sort of hissing sound. The length of hissing is proportional to the length of text string that it is given to speak. Since I'm running on a PPC system I fear the dreaded endian problem is to blame and that app_festival may need changing. Has anyone else experienced similar problems? Iain ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN messenger and *
Sorry for the late reply. I try port 5060 and it just knocks me back straight away, I cant see it even try to authenticate in the CLI. X-lite works both inside the LAN and outside using SIP. Messenger version = 4.7 John I will try your suggestion with sip.conf thanks for the help. I notice a few differences, I seem to be missing some bits.. Its like it is trying to authenticate with the Linux box and not asterisk. Sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=sip ; Default for incoming calls allow=ulaw allow=alaw allow=gsm allow=ilbc [3001] type=friend username=3001 fromuser=Craig1 secret=secret host=dynamic mailbox=3001 context=sip dtmfmode=info I found 3 guides and each one seems to be a bit different and use different ports. I am using the X100P, it is a home system, to reduce call charges for my family overseas. If I can get Messengger working it will be easier to talk them through the setup.
Re: [Asterisk-Users] Re: call files
Hi!
What I would like to get round this is probably the reverse I dont
want the people I am calling to hear ringing. For example as soon as it
Swap the numbers around.
I cannot figure this out - just swap them round?
But if I swap it round
Channel: SIP/User
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: internal
Extension: ???
Priority: 1
Try something like this:
Context: isdn_outgoing
Extension: 12345678
Priority: 1
[isdn_outgoing]
exten = .,1,Dial(CAPI/yourMSN:${EXTEN},,rT)
Cheers, Philipp
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[Asterisk-Users] Sipura 2000 configuration.
Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf [lcs-sipura1] context=main username=lcs-sipura1 secret= host=dynamic canreinvite=no nat=1 disallow=all allow=ulaw allow=alaw ; [lcs-sipura2] context=main username=lcs-sipura2 secret= host=dynamic canreinvite=no nat=1 disallow=all allow=ulaw allow=alaw Extensions.conf exten = 203,1,Dial(SIP/lcs-sipura1) exten = 204,1,Dial(SIP/lcs-sipura1) - \ \\_ Ariel Batista //IS Director / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 2000 configuration.
You have SIP/lcs-sipura1 listed for both extensions in your extensions.conf. Is this a type-o in your email? -Original Message- From: Ariel Batista [mailto:[EMAIL PROTECTED] Sent: Monday, December 22, 2003 1:11 PM To: Asterisk User List Subject: [Asterisk-Users] Sipura 2000 configuration. Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf [lcs-sipura1] context=main username=lcs-sipura1 secret= host=dynamic canreinvite=no nat=1 disallow=all allow=ulaw allow=alaw ; [lcs-sipura2] context=main username=lcs-sipura2 secret= host=dynamic canreinvite=no nat=1 disallow=all allow=ulaw allow=alaw Extensions.conf exten = 203,1,Dial(SIP/lcs-sipura1) exten = 204,1,Dial(SIP/lcs-sipura1) - \ \\_ Ariel Batista //IS Director / Avionica, Inc. -- [EMAIL PROTECTED] Ph: 786-544-1114 Fx: 305-574-0212 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 configuration.
exten = 203,1,Dial(SIP/lcs-sipura1) exten = 204,1,Dial(SIP/lcs-sipura1) dont you mean: exten = 203,1,Dial(SIP/lcs-sipura1) exten = 204,1,Dial(SIP/lcs-sipura2) ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audio format for announcements
1) Is it possible to store the menu sounds in wav ...sure, just put your 8kHz 16 bit mono files named whatever.wav in /var/lib/asterisk/sounds - asterisk will convert them to what is needed if needed. John This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicetronics
iTS [EMAIL PROTECTED] wrote: Thanks, any special configuration requirement? Nop, but you need to patch channel_vpb.c. Search the archives (oct - nov?) Jorge Tjapko. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza Sent: Lunes, 22 de Diciembre de 2003 05:01 p.m. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] voicetronics iTS [EMAIL PROTECTED] wrote: Hi list, Is there anyone using the voicetronics openline4 with asterisk. Does this card work ok for 4 port analoge fxo? Thanks, Tjapko. We are using openline4 for testing purpose, not in production. It seems to work. Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
I cracked the concept of how to handle incoming calls and route them to the right context, apologies for being a little slow on the uptake. I can now call between pots end points and netmeeting endpoints. Still having problems with sound despite having set everything to use G711A. POT to POT via * fine and netmeeting to netmeeting direct is OK. NM to NM via * is silent in both directions. NM to POT via * is silent NM to POT, but OK POT to NM i.e. the NM user can hear the POT user but not the other way. any pointers gratefully accepted. At 14:07 22/12/03, you wrote: I'm missing something here. I've put the following in extensions.conf and a few variations thereof. I've taken the sample configs and added to them, so when I dial 2200 from netmeeting * answers and runs me through the demo announcements. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as a PSTN gateway for SER
First off, here is what I want to do: SIP Clients - SER - Asterisk - VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients can talk to each other. I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: Here is the output: -- Executing Dial(SIP/-08114560, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/SIPprovider-5e0c is making progress passing it to SIP/-08114560 -- SIP/SIPprovider-5e0c answered SIP/-08114560 -- Attempting native bridge of SIP/-08114560 and SIP/SIPprovider-5e0c I have tried this with my SIP client behind a NAT and outside of a NAT, so I don't that is the problem. I have also tried this with both IAX and SIP providers and the problem is the same. One ring, and then silence. Any thoughts? Thank you for your time. __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN-PRI - WCT1XXP error
On Monday 22 of December 2003 17:54, Daniel Bichara wrote: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed your zaptel.conf is wrong it has to be: Follows my /etc/zaptel.conf: span=1,0,0,ccs,hdb3 wrong nethdlc=1-15 bchan=1-15 dchan=16 loadzone = us Thanks in advance, -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a PSTN gateway for SER
how did u setup your asterisk for this: I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 2:42 PM Subject: [Asterisk-Users] Asterisk as a PSTN gateway for SER First off, here is what I want to do: SIP Clients - SER - Asterisk - VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients can talk to each other. I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: Here is the output: -- Executing Dial(SIP/-08114560, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/SIPprovider-5e0c is making progress passing it to SIP/-08114560 -- SIP/SIPprovider-5e0c answered SIP/-08114560 -- Attempting native bridge of SIP/-08114560 and SIP/SIPprovider-5e0c I have tried this with my SIP client behind a NAT and outside of a NAT, so I don't that is the problem. I have also tried this with both IAX and SIP providers and the problem is the same. One ring, and then silence. Any thoughts? Thank you for your time. __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: Questions and finding
I installed * to primarily test its voicemail
feature. I installed it on a server WITHOUT any telco board (i.e.,
digium). Installation looks ok, however I am having problems.
MY SETUP:
2xATAs are configured to use * as
GkorProxy
Asterisk is registered to my SER SIP/RTP
Proxy
1.) First test
- ATA1 calls to ATA2. When voicemail starts
playing, it just cuts-off after 5-10seconds (consistently).
- Solution: I have to reconfigure ATA to use
OutboundProxy to be Asterisk IP.
- Am I doing the right thing?
2.) Second test:
- ATA1 calls ATA2 and left a message. * sends
an email to the owner of ATA2.
- ATA2 retrieves the message by dialing the
extension (i.e., exten=,1,VoiceMailMain(123))
- VoiceMail prompts for password. ATA2 keys
in its pasword. * doesn't reply and keeps on prompting for 5 times, until
it disconnects:
-- Executing
VoiceMailMain("SIP/6882332-6b3a", "123") in new stack --
Playing 'vm-password' (language 'en') -- Incorrect
password '' for user '123' (context = any) --
Playing 'vm-incorrect' (language 'en') -- Playing
'vm-password' (language 'en') -- Incorrect password '' for
user '123' (context = any) -- Playing
'vm-incorrect' (language 'en') -- Playing 'vm-password'
(language 'en')
- Tried to change AudioMode of ATA, to use
0x00140014, 0x11241124, and default 0x00150015. But there were no
difference.
- Tried to skip the password on *, using
exten=,1,VoiceMailMain(s123). * prompted there arenew
messages, and instructions on how to retreive them. Unfortunately,
none of the dtmf keys worked (1, #, etc).
Any solution to this one?
My thinking was that DTMF can only be detected by *
3.) How can I configure * to support the following
setup. I know * doesn't work 100% like softswitch, but maybe this type of
connection is supported?
ATA - * - CiscoAS5300
ATA to *, is IP connection
* to CiscoAS5300 is also IP connection
I appreciate your help. Thank
you.
Re: [Asterisk-Users] MSN messenger and *
Hi! I try port 5060 and it just knocks me back straight away, I cant see it even try to authenticate in the CLI. You won't see anything unless you type sip debug in the CLI. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToIP (TDD over IP)
On Mon, 22 Dec 2003, Philipp von Klitzing wrote: Excuse my ignorance: What exactly is TDD? Is it US specific? TDD - Telecommunications Device for the Deaf (also used by people with speech problems). Also known as a TTY (Telephone Typewriter) or TDY (not sure what it means) I don't know if it is US-specific or not. -- Joel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a PSTN gateway for SER
In sip.conf I have the following: context=OUTGOING autocreatepeer=yes [Provider] type=friend username=X secret=X host=x.FakeProvider.com So when Asterisk receives a call from SER it will autocreatepeer and give access to the OUTGOING context. --- Jess Magnaye [EMAIL PROTECTED] wrote: how did u setup your asterisk for this: I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: - Original Message - From: jerk face [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 2:42 PM Subject: [Asterisk-Users] Asterisk as a PSTN gateway for SER First off, here is what I want to do: SIP Clients - SER - Asterisk - VoIP provider Where SER will handle communications between SIP clients (since I would prefer that my SIP clients not use all of my bandwidth) Asterisk will handle calls to a VoIP provider I have read that people have similar setups working, but I have not seen any documentation of these setups. So far, SIP Clients can talk to each other. I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: Here is the output: -- Executing Dial(SIP/-08114560, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/SIPprovider-5e0c is making progress passing it to SIP/-08114560 -- SIP/SIPprovider-5e0c answered SIP/-08114560 -- Attempting native bridge of SIP/-08114560 and SIP/SIPprovider-5e0c I have tried this with my SIP client behind a NAT and outside of a NAT, so I don't that is the problem. I have also tried this with both IAX and SIP providers and the problem is the same. One ring, and then silence. Any thoughts? Thank you for your time. __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP Packet Time (20ms)
Hi, I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? This is causing problems with the Sipura SPA2000 on our network. The SPA does not like this and is treating many packets as lost packets (even though an Ethereal RTP Analysis trace shows they were not lost). Regards, Andres http://www.telesip.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * -- C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If that is the case, then what is in packet 52 and 55? There's not enough time between packets for 20ms of voice, unless it's repeating audio in the packets... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
I might be wrong, but isn't is just saying that the packet has been delayed x-ms? I'm not sure it's saying that Packet 52 arrived 5ms after packet 51. Although even if it was, that doesn't mean that it was sent 5ms after packet 51 either. -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Monday, December 22, 2003 3:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms) Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If that is the case, then what is in packet 52 and 55? There's not enough time between packets for 20ms of voice, unless it's repeating audio in the packets... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tor2 does not load
Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7 Did not get DONE signal. Short file maybe?? Registered Tormenta2 PCI ZT_SPANCONFIG failed on span 1: No such device or address (6) /lib/modules/2.4.18/misc/tor2.o: post-install tor2 failed /lib/modules/2.4.18/misc/tor2.o: insmod tor2 failed asterix:~# The module is listed by lsmod: asterix:~# lsmod Module Size Used byNot tainted tor2 84480 0 (unused) If I try to remove: asterix:~# rmmod tor2 Unable to handle kernel paging request at virtual address d08bc400 printing eip: d08a2c19 *pde = 0fdd4067 *pte = Oops: 0002 CPU:0 EIP:0010:[d08a2c19]Not tainted EFLAGS: 00010286 eax: d08bc000 ebx: cffe0c00 ecx: 6ea8 edx: d084cf40 esi: cef18000 edi: d08a2000 ebp: bfffecf8 esp: ce3fff48 ds: 0018 es: 0018 ss: 0018 Process rmmod (pid: 461, stackpage=ce3ff000) Stack: cffe0c00 d08b68a0 d08a2000 c01dbac4 0010 0282 c020b5dc c018ee5f cffe0c00 d08a2000 fff0 d08a4b40 d08b68a0 c020b488 0203 d08a2000 fffe ced51000 bfffecf8 c0114023 d08a2000 fff0 ced51000 bfffecf8 Call Trace: [d08b68a0] [c01dbac4] [c018ee5f] [d08a4b40] [d08b68a0] [c0114023] [c01134c7] [c0106b1b] Code: c6 80 00 04 00 00 00 8b 86 80 00 00 00 c6 80 01 04 00 00 00 Segmentation fault asterix:~# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If that is the case, then what is in packet 52 and 55? There's not enough time between packets for 20ms of voice, unless it's repeating audio in the packets... In this short example, if you add up all of the times shown and divide by the number of entries, you'll see its exactly 20ms of voice. The two delays of 50ms each are the problems (obviously). All that short trace really says is during the 120ms sampling interval, there was no dropped udp packets (since they nicely add up). If they didn't add up, it would be a problem. So, the question really is... what caused the two 50 ms delays? In technical network terms those delays are called jitter, and in the case noted above, that jitter is substantial. If that trace was taken next to the equipment (no routers involved in the middle), then which ever piece of equipment that originated those packets should be looked at carefully. If doesn't make any difference if anyone was talking for not; the packets are still going to flow, and they should be flowing at a very constant rate with reasonable but constant inter-packet delay (jitter). There really is nothing more that can be said without additional detail. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a PSTN gateway for SER
jerk face wrote: In sip.conf I have the following: context=OUTGOING autocreatepeer=yes [Provider] type=friend username=X secret=X host=x.FakeProvider.com So when Asterisk receives a call from SER it will autocreatepeer and give access to the OUTGOING context. Could you please explain autocreatepeer a bit more? I can't find any documentation on it. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: call files
I have tried this from the manager console and call files and it doesn't
seem to work the other way round. It will call the sip channel but not
the capi channel - in fact with capi debug this doesn't show anything
getting through
Asterisk monitor comes up twith
Attempting call on sip/nick ofr number@isdnout:1 (Retry 1)
Channel Sip/nick-dd98 was answered.
Then the sip call is dropped (by asterisk) then nothing!
Any others ideas???
Nick
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: 22 December 2003 18:17
To: Nick Knight
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Re: call files
Hi!
What I would like to get round this is probably the reverse I
don(tm)t
want the people I am calling to hear ringing. For example as soon as
it
Swap the numbers around.
I cannot figure this out - just swap them round?
But if I swap it round
Channel: SIP/User
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: internal
Extension: ???
Priority: 1
Try something like this:
Context: isdn_outgoing
Extension: 12345678
Priority: 1
[isdn_outgoing]
exten = .,1,Dial(CAPI/yourMSN:${EXTEN},,rT)
Cheers, Philipp
___
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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Monday 22 December 2003 15:36, Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * -- C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything. Thanks! Andres Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Monday 22 December 2003 15:55, Andrew Kohlsmith wrote: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If that is the case, then what is in packet 52 and 55? There's not enough time between packets for 20ms of voice, unless it's repeating audio in the packets... All packets contain 20ms of voice. Its just that they are not evenly spaced. My first guess will be to check out our traffic shaper, now that I know its not due to the Asterisk software. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DID trunks -- equipment requirement
Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1 correspondence. Since I'm in a less populated area of the country, SBC doesn't seem to have much in the way of fractional T1 products (on the scale that we need them) available, so I think my only option for DID is to use (analog) DID trunks for incoming calls and POTS lines for outbound calls. I'm familiar w/ POTS lines and have already done limited testing w/ a CAC channel bank equipped with FXO cards and that works fine. What I'm concerned about is the DID trunks. I've been told they have no dialtone and of course you can't place calls on them, but can receive calls. My question is, in general, should my CAC channel bank w/ the FXO cards that work on POTS lines work okay w/ analog DID trunks from the phone company? Might I have to purchase additional equipment to handle the DIDs (going into one of two Digium T1 cards I have in the Asterisk box)? Would they be different cards to plug into the CAC channel bank? Something totally different? Sorry to bring what I know is a rather off-topic question here, but the SBC guys don't like to help with customer education so much. As always, I appreciate all of your expertise and patience with me and the other new guys. John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tor2 does not load
On Mon, 2003-12-22 at 15:23, Eduardo Goncalves wrote: Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7 Did not get DONE signal. Short file maybe?? Just a guess, but maybe your module file is corrupted. Have you tried recompiling the module? If that doesn't work, try the standard move the card to a different slot. Sometimes cards can become belligerent and will not wake up until they have been initialized in a different slot. This is not a digium specific trick, but a problem I have had with other cards. Registered Tormenta2 PCI ZT_SPANCONFIG failed on span 1: No such device or address (6) /lib/modules/2.4.18/misc/tor2.o: post-install tor2 failed /lib/modules/2.4.18/misc/tor2.o: insmod tor2 failed asterix:~# The module is listed by lsmod: asterix:~# lsmod Module Size Used byNot tainted tor2 84480 0 (unused) If I try to remove: asterix:~# rmmod tor2 Unable to handle kernel paging request at virtual address d08bc400 printing eip: d08a2c19 *pde = 0fdd4067 *pte = Oops: 0002 CPU:0 EIP:0010:[d08a2c19]Not tainted EFLAGS: 00010286 eax: d08bc000 ebx: cffe0c00 ecx: 6ea8 edx: d084cf40 esi: cef18000 edi: d08a2000 ebp: bfffecf8 esp: ce3fff48 ds: 0018 es: 0018 ss: 0018 Process rmmod (pid: 461, stackpage=ce3ff000) Stack: cffe0c00 d08b68a0 d08a2000 c01dbac4 0010 0282 c020b5dc c018ee5f cffe0c00 d08a2000 fff0 d08a4b40 d08b68a0 c020b488 0203 d08a2000 fffe ced51000 bfffecf8 c0114023 d08a2000 fff0 ced51000 bfffecf8 Call Trace: [d08b68a0] [c01dbac4] [c018ee5f] [d08a4b40] [d08b68a0] [c0114023] [c01134c7] [c0106b1b] Code: c6 80 00 04 00 00 00 8b 86 80 00 00 00 c6 80 01 04 00 00 00 Segmentation fault asterix:~# ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sweet video phone
Supports H323 http://www.viseon.com/prod/c_VisiFone.asp?id=133
Re: [Asterisk-Users] Sweet video phone
E h323 evil. On Mon, 22 Dec 2003, Steve Totaro wrote: Supports H323 http://www.viseon.com/prod/c_VisiFone.asp?id=133 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DID trunks -- equipment requirement
On Monday, December 22, 2003 3:40 PM, john lawler [SMTP:[EMAIL PROTECTED] wrote: Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1 correspondence. Since I'm in a less populated area of the country, SBC doesn't seem to have much in the way of fractional T1 products (on the scale that we need them) available, Have you asked for a full T1 but with just 10-14 DID/DOD trunks? We can not get fractional T1 here but on a full T1 we can add anywhere from 1 - 24 trunks. So we pay one amount per month for the T1 and on top of that we pay another amount times the number of trunks we have. I know this didn't exactly address your questions. For your primary question I believe that your would need different type of channels in a channel bank than FXOs. DPTs (Dial pulse) terminating come to mind, but that may be wrong. Don Pobanz so I think my only option for DID is to use (analog) DID trunks for incoming calls and POTS lines for outbound calls. I'm familiar w/ POTS lines and have already done limited testing w/ a CAC channel bank equipped with FXO cards and that works fine. What I'm concerned about is the DID trunks. I've been told they have no dialtone and of course you can't place calls on them, but can receive calls. My question is, in general, should my CAC channel bank w/ the FXO cards that work on POTS lines work okay w/ analog DID trunks from the phone company? Might I have to purchase additional equipment to handle the DIDs (going into one of two Digium T1 cards I have in the Asterisk box)? Would they be different cards to plug into the CAC channel bank? Something totally different? Sorry to bring what I know is a rather off-topic question here, but the SBC guys don't like to help with customer education so much. As always, I appreciate all of your expertise and patience with me and the other new guys. John Lawler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a PSTN gateway for SER
autocreatepeer I just found out about this today from the Asterisk-Dev mailing list. The email was from John Bigelow and is as follows: This will allow any sip user to register with asterisk with no authentication. So if you are lazy or for whatever reason do not want to create the peers in the sip.conf you can set autocreatepeer=yes in sip.conf and anyone can place calls through the system. -John --- Olle E. Johansson [EMAIL PROTECTED] wrote: jerk face wrote: In sip.conf I have the following: context=OUTGOING autocreatepeer=yes [Provider] type=friend username=X secret=X host=x.FakeProvider.com So when Asterisk receives a call from SER it will autocreatepeer and give access to the OUTGOING context. Could you please explain autocreatepeer a bit more? I can't find any documentation on it. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? New Yahoo! Photos - easier uploading and sharing. http://photos.yahoo.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Monday 22 December 2003 16:37, Andres wrote: On Monday 22 December 2003 15:36, Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * -- C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything. Thanks! Andres Ok...after some more testing, the traffic shaping software was not the culprit. It turns out that if the UA is configured for 60ms of voice, then Asterisk will show this strange behaviour. If we set the UA for 20ms, then all works well. Thanks. Andres Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Monday 22 December 2003 16:37, Andres wrote: On Monday 22 December 2003 15:36, Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * -- C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything. Thanks! Andres Ok...after some more testing, the traffic shaping software was not the culprit. It turns out that if the UA is configured for 60ms of voice, then Asterisk will show this strange behaviour. If we set the UA for 20ms, then all works well. Cool! How did it get set to 60ms? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers - disappearing
I think we've having some luck with this setting. Of course we had to crank it up higher so that it didn't consider the clients LAGGED. When the clients were LAGGED they couldn't receive any calls for some reason. So like a setting of 200ms seems to work fine for everyone. Eric Wieling wrote: Their firewall may be timeing them out. Try adding qualify=60 to each of the entries in sip.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile Problem
Sorry for the dup post but never got a reply so I am reposting below: I am trying to compile the asterisk and if fails at the end on: make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx'gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread`/usr/lib/gcc-lib/i486-slackware-linux/3.3.2/../../../../i486-slackware-linux/bin/ld: cannot find -lXextcollect2: ld returned 1 exit statusmake[1]: *** [pbx_gtkconsole.so] Error 1make[1]: Leaving directory `/usr/src/asterisk-0.5.0/pbx'make: *** [subdirs] Error 1 Anyone know what is wrong? Linpri and zaptel compiled just fine. This is linux slackware 2.4.23 all the latest from 9.1 slackware distrib fresh system install. Thanks, John
Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
On Monday 22 December 2003 19:58, Rich Adamson wrote: On Monday 22 December 2003 16:37, Andres wrote: On Monday 22 December 2003 15:36, Rich Adamson wrote: I have a question regarding the Asterisk Packet Time for SIP Calls. It is hardcoded at 20ms but when I do an RTP Analysis on a stream it is clear that these packets are not spaced out at 20ms. In general you see something like: Packet 50 - Delay 50ms Packet 51 - Delay 5ms Packet 52 - Delay 5ms Packet 53 - Delay 50ms Packet 54 - Delay 5ms Packet 55 - Delay 5ms Is there anyway to space them out evenly at 20ms?? The 20 ms is not the inter-packet timing, its the relative content of what's within the packet. In other words, the packet contains 20ms of encoded voice. If the inter-packet times (delays) are large, as they would seem to be in your example, then something else is not right. Possibly a half-duplex ethernet connection, something else running on the server, router buffers, etc. On a typical * -- C7960 local call, I generally see from 1ms to 20ms inter-packet delays. Seldom (if ever) anything above 20ms. Thanks for your Input Rich. I went ahead and tested this on our production servers and sure enough the inter-packet times are 20ms. There must be something happening with our LAB Asterisk. It could be the CBQ traffic shaping software we have running on it. I will fiddle around with it to see if it changes anything. Thanks! Andres Ok...after some more testing, the traffic shaping software was not the culprit. It turns out that if the UA is configured for 60ms of voice, then Asterisk will show this strange behaviour. If we set the UA for 20ms, then all works well. Cool! How did it get set to 60ms? The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the transmit packet size to 60ms (or multiple other values). Asterisk will receive 60ms and transmit 20ms times 3 packets, andit works quite well. In any case our SPA2000 problem was unrelated to the packet time. Regards, Andres ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compile Problem
Hi, On Tue, 2003-12-23 at 12:12, [EMAIL PROTECTED] wrote: [...] I am trying to compile the asterisk and if fails at the end on: make[1]: Entering directory `/usr/src/asterisk-0.5.0/pbx' gcc -shared -Xlinker -x -o pbx_gtkconsole.so pbx_gtkconsole.o `gtk-config --libs gthread` /usr/lib/gcc-lib/i486-slackware-linux/3.3.2/../../../../i486-slackware-linux/bin/ld: cannot find -lXext collect2: ld returned 1 exit status make[1]: *** [pbx_gtkconsole.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk-0.5.0/pbx' make: *** [subdirs] Error 1 Anyone know what is wrong? Linpri and zaptel compiled just fine. This is linux slackware 2.4.23 all the latest from 9.1 slackware distrib fresh system install. I'm not sure where you'd find the following file in Slackware, but in RedHat: /usr/X11R6/lib/libXext.so.6 .. is part of the XFree86-libs RPM. Find the corresponding tgz, install it and then try to compile again. It should get past that error. HTH, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MSN messenger and *
use this [3001] type=friend ;username=3001 ;fromuser=Craig1 ;secret=secret host=dynamic mailbox=3001 context=sip dtmfmode=info auth=plaintext make sure ur MSN version is 4.7.0105. -B - Original Message - From: Craig Waddington To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 10:10 AM Subject: [Asterisk-Users] MSN messenger and * Sorry for the late reply. I try port 5060 and it just knocks me back straight away, I cant see it even try to authenticate in the CLI. X-lite works both inside the LAN and outside using SIP. Messenger version = 4.7 John I will try your suggestion with sip.conf thanks for the help. I notice a few differences, I seem to be missing some bits.. Its like it is trying to authenticate with the Linux box and not asterisk. Sip.conf [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind to context=sip ; Default for incoming calls allow=ulaw allow=alaw allow=gsm allow=ilbc [3001] type=friend username=3001 fromuser=Craig1 secret=secret host=dynamic mailbox=3001 context=sip dtmfmode=info I found 3 guides and each one seems to be a bit different and use different ports. I am using the X100P, it is a home system, to reduce call charges for my family overseas. If I can get Messengger working it will be easier to talk them through the setup. Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
Re: [Asterisk-Users] ToIP (TDD over IP)
Tilghman Lesher wrote: On Monday 22 December 2003 10:12, Philipp von Klitzing wrote: Hi! I'm also curious if anyone else is doing this or if anyone else is using the Asterisk TDD support. Excuse my ignorance: What exactly is TDD? Is it US specific? It's a specification for sending words over a normal telephone, normally used by the deaf. It resembles the old-style modems in that the handset is interfaced with a microphone and speaker. This allows TDD to be used with payphones, which do not have an RJ-11 interface. -Tilghman Does anyone use this any more? All the deaf people here carry cellphones, and use SMS to communicate. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the bandwidth requirement for IAX
Hello, I am trying to figure out how much bandwidth asterisk requires using IAX between 2 boxes if all available channels are used. Scenarios: A. 1 TE410P---Asterisk A --- Internet --- Asterisk B--1 TE410P B. 2 TE410P---Asterisk A --- Internet --- Asterisk B--2 TE410P C. 3 TE410P---Asterisk A --- Internet --- Asterisk B--3 TE410P Thanks in advance. Allan G. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sweet video phone
Steve Totaro wrote: Supports H323 http://www.viseon.com/prod/c_VisiFone.asp?id=133 So? Whilst there are still only a few VoIP audio phones available, almost every computer related manufacturer in Asia has at least one video phone model like this. There must be dozens of units like this available right now. Usually they use H323. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX trunking recomendations
Hi, Im conecting to * servers using IAX2 no NAT in my setup. I read Wikis docs and lists archives but there is no recomendation about what to use. If I understand I can use friend in both sides but its isnt recomended; so I should define both a peer and a user on each box ? cause AFAIR in a trunking enviroment both box place and receive calls. Do I need to use register ? Please take in account that Ill like to ease provisioning and avoid thinks like Dial(IAX2/user:[EMAIL PROTECTED]/exten) and I noted if the remote box is registered I can just Dial(IAX2/box/exten). Also I noted theres a switch object there isnt much docs about do I need boxes registered between themselves to use switch ? TIA -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sweet video phone
They recently announced SIP. See: http://www.voip-info.org/wiki-Viseon+VisiFone Anyone know where to actually buy one? Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 22, 2003 12:31 PM Subject: Re: [Asterisk-Users] Sweet video phone E h323 evil. On Mon, 22 Dec 2003, Steve Totaro wrote: Supports H323 http://www.viseon.com/prod/c_VisiFone.asp?id=133 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MSN to GS - Call drops in 10 secs
Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general]port = 5060bindaddr = 0.0.0.0context = bogon-calls;context = defaultdisallow=allallow=ulawallow=alawallow=ilbcallow=gsm ;My SIP phone - GS[2000]type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband ;MSN Msgr[2002]type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji Do you Yahoo!? Yahoo! Photos - Get your photo on the big screen in Times Square
Re: [Asterisk-Users] Compile Problem
/usr/X11R6/lib/libXext.so.6 .. is part of the XFree86-libs RPM. Find the corresponding tgz, install it and then try to compile again. It should get past that error. That did it thanks! John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ToIP (TDD over IP)
It's a specification for sending words over a normal telephone, normally used by the deaf. It resembles the old-style modems in that the handset is interfaced with a microphone and speaker. This allows TDD to be used with payphones, which do not have an RJ-11 interface. Does anyone use this any more? All the deaf people here carry cellphones, and use SMS to communicate. We have a local company that contracts with several States to provide TDD services, and they just implemented video 323 to allow folks to sign (visual TDD) over the Internet. Still a big deal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] First VOIP *
Hello, I am setting up a VOIP system using * for our remote located broadband customers. We are bringing in a full voice T1 with 24 channels and going to use the wildcard T100P. Its going to be another 2 weeks before our voice T1 is installed and I want to take that time to setup our * box. Its all installed and working right now and I just need to do configuration. Our goal is to do a vonage type service but to our customers already as a value added service. We will be assigning each customer there own DID telephone number. They will be using something like a cisco ip phone, still deciding on this part - Any recomendations on this for this type of application? Anyhow, the standard config examples show basic setups that are geared more toward office setups etc.. Is there an example config on how to setup for a VOIP server serving remote clients? I want to just do some simple things here. 1) Assign a DID phone number to a customer based on static IP. 2) (INCOMING) When a land line voice call comes in to customer DID 555-555- it is recognized as belonging to 10.0.0.10 by * and is forwarded to that ip's phone. 3) (OUTGOING) When user 10.0.0.10 picks up his ip phone he gets a dial tone at the * box via his broadband connection and dials a land line and the * b ox will just pick the next available channel out of the bunch and give him outbound land call without dialing 9. Repeat for next customer... Can someone point me to some docs or example configs on this please!? Many thanks in advance, John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wishes
I want wish You all - asterisk-men - Merry Christmas and excellent New Year ! Radoslaw --- Let's talk about what connecting us together (c) Lafinion [RW] http://www.lafinion.com mailto:[EMAIL PROTECTED] : MSN: [EMAIL PROTECTED] Yahoo ID: Lafinion ICQ: 323220515 : : AIM: Lafinion GG: 3282800 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
