Re: [Asterisk-Users] dialing codes..( You can help! )
It looks like its going to be a bigger job than I thought it would be.. I guess I will have to start with the countries that we use the most and then add the others as I find the details.. Thanks for all the input.. I'd be up for setting up some kind of website/database thing for collating all this information, just not sure of the value and if anyone else would be up for it/contributing data? Be cool to have though, and nice for customer bill presentation etc? Try www.numberingplans.com or www.numberplan.org - they are both commerical but have some information for free. Also you can look at http://www.wtng.info/ a free site, and the ITU site at http://www.itu.int/ITU-T/inr/nnp/index.html its all there if you know where to look! Linus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App_festival crashing
Hi.
I am not using cache, just :
festival.conf
-
[general]
host=localhost
port=1314
festivalcommand=(tts_textasterisk %s 'file)(quit)\n
but in extensions.conf when i call the festival app i put the text
'quoted' like this:
exten = 003,1,Festival('Hello asterisk user, how are you today?') ;
-- note the quotes ...
exten = 003,2,Wait(1)
exten = 003,3,Hangup()
And everything works like the readme says.
Good luck.
Borut Senicar wrote:
Hi all,
I'm unable to put app_festival to work. I successfully patched,
installed and tested festival (interactive logon and telnet to server
port) which seems to work without problems.
But when I test it in asterisk I got the following trace in console:
-- Executing Answer(SIP/bsenicar-850b, ) in new stack
-- Executing SayDigits(SIP/bsenicar-850b, 123) in new stack
-- Playing 'digits/1'
-- Playing 'digits/2'
-- Playing 'digits/3'
-- Executing Festival(SIP/bsenicar-850b, Connect to Festival) in
new stack
== Parsing '/etc/asterisk/festival.conf': Found
WARNING[147466]: File app_festival.c, Line 304 (festival_exec): Text
passed to festival server : Connect to Festival
WARNING[147466]: File app_festival.c, Line 353 (festival_exec): line
length : 19
WARNING[147466]: File app_festival.c, Line 357 (festival_exec): Seek
position : 23
WARNING[147466]: File app_festival.c, Line 381 (festival_exec): Passing
text to festival...
WARNING[147466]: File app_festival.c, Line 390 (festival_exec): Writing
result to cache...
WARNING[147466]: File app_festival.c, Line 400 (festival_exec): Passing
data to channel...
== Spawn extension (home-trusted, 1000, 3) exited non-zero on
'SIP/bsenicar-850b'
In festival.conf I enabled all 5 default options and my extensions.conf
looks like this:
[home-trusted]
exten = 1000,1,answer
exten = 1000,2,SayDigits(123)
exten = 1000,3,Festival(Connect to Festival)
exten = 1000,4,Wait(5)
exten = 1000,5,Festival(send the argument)
exten = 1000,6,Hangup
Cache file is created but playback to channel doesn't work correctly.
I'm running Asterisk CVS-09/23/03-23:16:24
I also noticed that parsing of festival.conf in app_festival.c is done
incorrectly for usecache.
On line 281 of app_festival.c
usecache = ast_true(temp);
value of usecache config entry is tested with ast_true function, which
returns -1 if value is (yes, y, t or 1). For that reason cache is never
used.
Correct line should be:
usecache = ast_true(temp)==-1;
Thanks in advance.
Borut
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Re: [Asterisk-Users] list of voice prompts
Take a look at sounds.txt in the root of your Asterisk source.. Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list of voice prompts
Me = stupid!!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: woensdag 24 september 2003 11:22 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] list of voice prompts Take a look at sounds.txt in the root of your Asterisk source.. Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 23 September 2003 19:04, jerk face wrote: I keep getting segmentation faults when I do a reload. Do what Critchfield and Pycko told you to do, but I'm betting you're using chan_h323 with a gatekeeper (it's a known bug I think). That's what's killing my Asterisk on reload: (gdb) bt #0 0x40571b3e in PSocket::os_select(int, fd_set*, fd_set*, fd_set*, PScalarArrayint const, PTimeInterval const) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #1 0x4056aef2 in PSocket::Select(PSocket::SelectList, PSocket::SelectList, PSocket::SelectList, PTimeInterval const) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #2 0x4056a8c5 in PSocket::Select(PSocket::SelectList, PTimeInterval const) () from /usr/lib/libpt_linux_x86_r.so.1.5.2 #3 0x40c380e7 in H323TransportUDP::DiscoverGatekeeper(H323Gatekeeper, H323RasPDU, H323TransportAddress const) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #4 0x40c48ebf in H323Gatekeeper::StartDiscovery(H323TransportAddress const) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #5 0x40c48cfc in H323Gatekeeper::DiscoverByAddress(H323TransportAddress const) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #6 0x40bfd59c in H323EndPoint::SetGatekeeper(PString const, H323Transport*) () from /usr/lib/libh323_linux_x86_r.so.1.12.2 #7 0x403ac4d4 in h323_set_gk (gatekeeper_discover=0, gatekeeper=0x403b4dc0 195.135.216.2, secret=0x403b4e40 ) at ast_h323.cpp:1005 #8 0x403a5231 in reload () at chan_h323.c:1643 #9 0x080554c2 in ast_module_reload () at loader.c:159 #10 0x0806d10a in handle_reload (fd=153, argc=1, argv=0xbd9ff61c) at cli.c:105 #11 0x0806cefa in ast_cli_command (fd=153, s=0x0) at cli.c:1006 #12 0x08085770 in netconsole (vconsole=0x80bcc28) at asterisk.c:193 #13 0x40023463 in pthread_detach () from /lib/libpthread.so.0 - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cWlx2TEAILET3McRAhjnAKCCJI93ty4OxpGZwIMNlchRaWiiTwCeNk/Y HVqmHi76DIZsofmMF46Bp1w= =gIC7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PROBLEMS WITH IAXATEL AND DIGIUM IAX
Hi I'm having a extrange problem I cant register with Iaxtel or call to digium... But i cant make or recive IAX calls... ( I made some one with irc users ) Any idea why? Alvaro, Have you moved your [iaxtel] user settings so as they are at the bottom of your iax.conf file? It's a known bug with IAXTel configuration on Asterisk where iaxtel attempts to read the last user/friend entry in the iax.conf file. HTH, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does SIP work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cXDg2TEAILET3McRAlWgAJ4/2Y21JU5VkfrO2CMLAMAfOdiszACgk/Yf lCKOXryUGv1nQOevTry+rqc= =pJJR -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meridian Option 11 and asterisk
Has anyone ever interfaced a merdian option 11 and asterisk. Just wondering how you went about, it's for a small setup me only need between 4/6 channels, I was thinking about using some spare ISDN channels between the two. Has anyone seen an SIP option for the meridian? European Museum Of The Year 2002 The Chester Beatty Library http://www.cbl.ie/ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] RE: [Asterisk-Users] can´t call ICH
Hi!
There is my sip.conf:
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
maxexpirey=180 ; Max length of incoming registration we allow
defaultexpirey=160 ; Default length of incoming/outoing registration
disallow=all
allow=gsm
allow=ulaw
allow=alaw
tos=reliability
register =user:[EMAIL PROTECTED]/33
register =user:[EMAIL PROTECTED]/33
register =user:[EMAIL PROTECTED]/33
[fwd]
type=friend
secret=ipfone001
username=400277
host=fwd.pulver.com
context=from-sip
[welcome]
type=friend
secret=welcome
username=5
host=fwd.pulver.com
context=from-sip
[iconnect]
type=friend
secret=3587
username=31451543
host=sipauth.deltathree.com
dtmfmode=inband
context=from-sip
[33]
type=friend
secret=33
username=33
host=dynamic
defaultip=192.168.0.31
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=snom200 33
[34]
type=friend
secret=34
username=34
host=dynamic
defaultip=192.168.0.36
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=snom100 34
[35]
type=friend
secret=35
username=35
host=dynamic
defaultip=192.168.0.33
dtmfmode=rfc2833
mailbox=331
context=from-sip
callerid=ipdialog 35
- Original Message -
From: Paul Crick [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, September 23, 2003 7:19 PM
Subject: [Asterisk-Users] RE: [Asterisk-Users] can´t call ICH
You've got a whole bunch of numbers you're trying to call there. What is
the
full number that you want to call including the country code? It's not
clear
if the number you're trying should be 755xxx 55xxx or 055xx ?
and my extensions:
[from-sip]
exten =33,1,DIAL(SIP/33,20,tr)
exten =34,1,DIAL(SIP/34,20,tr)
exten =35,1,DIAL(SIP/35,20,tr)
exten = _9x.,1,DIAL,Zap/g1/${EXTEN:1}
exten = _8X.,1,DIAL(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten =331,1,VoicemailMain,s331
The number i want to call is 55 11 36752312
I hope this helps.
Thanks !
Miklos
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[Asterisk-Users] CPU Optimisations For asterisk
How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? Thanks for your Help Robb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meridian Option 11 and asterisk
I have a cisco router connected to the Meridian with an E1 QSIG line. The router converts the meridian calls to SIP and forwards them to asterisk (and vice-versa). It works really well, but there are probably cheaper ways for a small setup - Perhaps put an analogue card in the Asterisk server and connect it to some of the meridian's analogue lines? http://www.psionic.com/application/pdf/en/us/guest/products/ps278/c1237/ccmi gration_09186a00800e0dd6.pdf -Original Message- From: Stephen Farrell [mailto:[EMAIL PROTECTED] Sent: 24 September 2003 11:03 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11 and asterisk Has anyone ever interfaced a merdian option 11 and asterisk. Just wondering how you went about, it's for a small setup me only need between 4/6 channels, I was thinking about using some spare ISDN channels between the two. Has anyone seen an SIP option for the meridian? European Museum Of The Year 2002 The Chester Beatty Library http://www.cbl.ie/ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. ___ 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] Does SIP work?
Tais M. Hansen schrieb: ... Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. ... Hi, we have 2 snom phones running with sip. (Asterisk-0.5.0). The sip part seems to be very stable. We have currently some problems with chan_modem (i4l). We are going to replace the isdn card by a avm one in order to use chan_capi. According to a lot of reports within this mailing list, we'll get rid of our remaining asterisk problems doing this. But chan_sip is very reliable in our environment. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Does SIP work?
*This message was transferred with a trial version of CommuniGate(tm) Pro* SIP Works fine. I use it every day. Check your config. What errors are you getting when you endpoints try to register? Also, go through the mailing list archives as there are sample configs in there somewhere. J -Original Message- From: Tais M. Hansen [mailto:[EMAIL PROTECTED] Sent: Wednesday, 24 September 2003 8:25 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Does SIP work? *This message was transferred with a trial version of CommuniGate(tm) Pro* -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cXDg2TEAILET3McRAlWgAJ4/2Y21JU5VkfrO2CMLAMAfOdiszACgk/Yf lCKOXryUGv1nQOevTry+rqc= =pJJR -END PGP SIGNATURE- ___ 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] CPU Optimisations For asterisk
Robert Boardman wrote: How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? CHOST=i686-pc-linux-gnu CFLAGS=-mcpu=athlon-xp -O3 -pipe Well, it might run slightly faster, but you probably won't really notice the difference. You might well be better off with -O2 rather than -O3, as -O3 tends to agressively unroll branches to inlines which reduces the amount of code that fits on the chip's cache, resulting in slowness. It's swings and roundabouts, really. If you're using echo cancelling, it should be quicker if you enable the MMX stuff for that (see the Asterisk Makefile). Why do you need the extra speed? If you're desperately trying to optimise things like this to gain extra performance, you must have a pretty big system. Pretty big system should mean you have the cash to upgrade your CPU a bit, which will make much more difference. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: list of voice prompts
Michiel Betel [EMAIL PROTECTED] said: Does there exist a text file with all the 'standard' Asterisk voice messages? I'm planning to get them recorded in dutch, but need to know the exact text of each prompt... Michiel - are you planning to release the recordings? I had the idea to to the same thing as Digium with the english sounds, get a base set released and have an option for people to order additional ones (more in order to help the 'voice' at hand make a bit of side money than to get rich ;-)). However, if you plan to release them, one set should be enough... -- Cees de Groot http://www.cdegroot.com [EMAIL PROTECTED] GnuPG 1024D/E0989E8B 0016 F679 F38D 5946 4ECD 1986 F303 937F E098 9E8B Cogito ergo evigilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CPU Optimisations For asterisk
Thanks for your reply The system is a small installation but I was thinking about optimizations and wondered if there would be any particular benifit anyway thanks for the reply, your comments are very useful robb Quoting Alastair Maw [EMAIL PROTECTED]: Robert Boardman wrote: How would I compile asterisk for the Athlon XP arch, would there be any advantage doing this? CHOST=i686-pc-linux-gnu CFLAGS=-mcpu=athlon-xp -O3 -pipe Well, it might run slightly faster, but you probably won't really notice the difference. You might well be better off with -O2 rather than -O3, as -O3 tends to agressively unroll branches to inlines which reduces the amount of code that fits on the chip's cache, resulting in slowness. It's swings and roundabouts, really. If you're using echo cancelling, it should be quicker if you enable the MMX stuff for that (see the Asterisk Makefile). Why do you need the extra speed? If you're desperately trying to optimise things like this to gain extra performance, you must have a pretty big system. Pretty big system should mean you have the cash to upgrade your CPU a bit, which will make much more difference. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.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] SIP phone hangs after some hours
Hi, I have a problem with sip.conf. After some hours my sip phone(netergy) hangs. In clonse appears the next logs repeatly: 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.155 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80 From: asterisk sip:[EMAIL PROTECTED];tag=as4b104f64 To: sip:192.168.0.155 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.155:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as4b104f64 To: sip:192.168.0.155 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80 Supported: timer,100rel Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 13 headers, 0 lines DEBUG[12301]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' My sip.conf is the next: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.207; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay maxexpirey=10 ; Max length of incoming registration we allow defaultexpirey=10 ; Default length of incoming/outoing registration [705] type=friend username=705 host=192.168.0.155 dtmfmode=inband mailbox=705 callerid=705 context=outgoing reinvite=yes canreinvite=no qualify=yes nat=-1 My sip phone doesn't register in asterisk due to my decision. I can send and receive call, but if phones is inactive during some hours it hangs. It is due to asterisk or my sip phone? Any idea? Thanks, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does SIP work?
Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. I use SIP to talk to Grandstream 100s every day, and also to the FWD network without issue. Are you trying to access SIP across NAT or other restrictive firewall or something? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP / GrandStream Configuration
Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf --- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a "register" statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel
Re: [Asterisk-Users] Does SIP work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 September 2003 13:42, Jamie Carl wrote: SIP Works fine. I use it every day. Check your config. What errors are you getting when you endpoints try to register? 401 Unauthorized Also, go through the mailing list archives as there are sample configs in there somewhere. I did, but none of the examples nor where the default config clear on the fact that the [xxx] HAD to match sip:[EMAIL PROTECTED]. Once that was cleared up, it started working for me too. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cZPH2TEAILET3McRArZKAJ4hgOIagGS/2FFinHsC395/OIXDUwCfSjNl 757RlrzQtj/tdSqj3FdTJoc= =wWg+ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does SIP work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 September 2003 14:38, Andrew Kohlsmith wrote: Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. I use SIP to talk to Grandstream 100s every day, and also to the FWD network without issue. Are you trying to access SIP across NAT or other restrictive firewall or something? No firewall, no nat. Only lack of documentation for sip.conf... I got it to work now. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cZQG2TEAILET3McRAu9MAJ9dGA6BVyTW/OBem/FZnzz1xBY9KACfbRge ILS+9IptUCB6HrsDDLVMrCA= =lRx5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
Try adding nat=yes to your config.. Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no.. Finally set dtmfmode=info for the GS phones.. Later.. Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf --- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a register statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using Asterisk in an netted scenario
Title: Mensaje Yes yo can do it. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de e-smithEnviado el: miércoles, 24 de septiembre de 2003 15:02Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Using Asterisk in an netted scenario Hi, Just to get myideeas confirmed: Is it possible to useasterisk in a scenario where : - One Asterisk connects to another asterisk over tcp/ip with qos to another asterisk.- The otherasterisk has an connection to the PSTN whitch users connected to the first asterisk uses to get to the public telephone network. Kind regards Mats Karlsson
Re: [Asterisk-Users] False RING (incoming call) on Digium X101P FXO
I was surprised to see that it's 240 volts (peak-to-peak)! Egad.. no wonder it shocks fingertips. 20Hz (50ms cycle), 2 second long clean sine waveform. I was just surprised to see twice as much voltage as expected. 240 sounds like a lot. Are you sure you were doing DC measurement? Normal is +/- 90 - 120, generally at least 48VRMS or about 70V. The ring generators in the central office are typically set for something between 90 - 120 volts AC (RMS, not peak-to-peak) and have been since the hand crank phones disappeared. DC voltage has never been used in the US. Those telco's with long rural subscriber loops are likely to be towards the upper end of the ring voltage, while metro offices are more likely to be towards the middle to lower end. Depending upon the age of the central office, the actual ring voltage will vary depending upon exactly how many phone lines are being rung at exact same time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Festival Problems
I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say trying and then hangup... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Hello 1010 Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... Sep 24 23:05:33 WARNING[606231]: File app_festival.c, Line 410 (festival_exec): Festival WV command Any ideas? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
Ok, here is the real gdb output. This GDB was configured as i586-mandrake-linux-gnu... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. --- Martin Pycko [EMAIL PROTECTED] wrote: actually gdb /usr/sbin/asterisk core.6044, sorry On Tue, 23 Sep 2003, jerk face wrote: I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400p loading errors
I just received one yesterday and it seems to work fine... I'm running RH9 w/SIP, IAX, and Zaptel devices. -ben On Monday, September 15, 2003, at 01:56 PM, Bob Knight wrote: I finally received a phone call from Silicon Labs. They left a voice mail saying they were going to email me a data sheet for Si3210. I have not received it yet. As soon as I do and I get a little free time I will kick the chip around a little and try to narrow down the problem. A few questions: 1. Has anyone received a new (since sept 1) tdm400p card that works? 2. Why isn't digium looking into this? OK. Now it is time for me to go back to my full time job of trying to find a job. Azher Amin wrote: Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed ProSLIC on module 3 insane (1) 0 should be 2 Module 3: Not installed /lib/modules/2.4.20-8/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-8/misc/wcfxs.o: insmod /lib/modules/2.4.20-8/misc/wcfxs.o failed /lib/modules/2.4.20-8/misc/wcfxs.o: insmod wcfxs failed I have another TDM400P revision C (few months older) which works perfectly on the same slot of the system. The machine is AMD750 and I have tested several other cards and they worked fine. Plz suggest me about this problem and how to correct it. Regards Azher -- -- Do you Yahoo!? Yahoo! SiteBuilder http://us.rd.yahoo.com/evt=10469/*http://sitebuilder.yahoo.com - Free, easy-to-use web site design software -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 2600 and ASTERISK
Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart
[Asterisk-Users] X100P incoming calls - hangup delay
Hi All, I wasn't too sure how to word the subject, so I apologise for that. Anyway, I've got two X100P cards here accepting calls. Basically in Australia our ring cadence is 400ms on, 200ms off, 400ms on, 2000ms off, repeat. What I've noticed is that it takes about 8 seconds after the caller has hungup for the internal phones to stop ringing, whereas in Australia the maximum time would be 2 seconds (or 2.5 to be safe). I was just wondering if somebody might know if there is somewhere in the source where I can reduce this value. Unfortunately my C knowledge is very limited - I could probably find it given time, but if somebody might be able to give me a bump in the right direction that would be greatly appreciated. Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
On Wed, 2003-09-24 at 15:41, jerk face wrote: Ok, here is the real gdb output. This GDB was configured as i586-mandrake-linux-gnu... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. I started experiencing segfaults a while back too. Both after a reload and during startup when loading res_adsi.so. What fixed it for me was changing -O6 in all Makefiles to -O2. I'm using Red Hat 9 and it has been said that Red Hat does some funky stuff with their gcc so that may be causing it to segfault when compiled with -O6. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] netconsole - bad file descriptor?
In looking through the /var/log/asterisk/messages log, I see about a million lines like: Sep 21 12:00:59 WARNING[1235176752]: File asterisk.c, Line 183 (netconsole): sel ect returned 0: Bad file descriptor That was about the time I was attempting to test overhead paging using: ; the following provides 699 as a paging system (speaker on console) exten = 699,1,Dial(CONSOLE/dsp) exten = 699,2,Hangup Anyone know if this is a known problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 200 errors?
The following error messages were observed in /var/log/asterisk/messages: Sep 22 10:26:42 NOTICE[1133735216]: File chan_sip.c, Line 5099 (handle_request): Unknown SIP command 'PUBLISH' from '212.23.220.236' Sep 22 11:32:50 WARNING[1133735216]: File chan_sip.c, Line 4519 (handle_response ): Got 200 OK on REGISTER that isn't a register The phone was a Snom 200 running v2.01t code. The phone has been working fine. Are these misconfigurations on my part or something that I should be concerned about? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
On Wed, 2003-09-24 at 08:41, jerk face wrote: Ok, here is the real gdb output. This GDB was configured as i586-mandrake-linux-gnu... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. This is where you type 'bt' and find out how it made it to that frame. --- Martin Pycko [EMAIL PROTECTED] wrote: actually gdb /usr/sbin/asterisk core.6044, sorry On Tue, 23 Sep 2003, jerk face wrote: I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
Title: Message I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz JozwiakSent: Wednesday, September 24, 2003 9:46 AMTo: ASTERISK USERSSubject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
I am running Mandrake 9.1 if that makes a difference. --- Patrick [EMAIL PROTECTED] wrote: On Wed, 2003-09-24 at 15:41, jerk face wrote: Ok, here is the real gdb output. This GDB was configured as i586-mandrake-linux-gnu... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. I started experiencing segfaults a while back too. Both after a reload and during startup when loading res_adsi.so. What fixed it for me was changing -O6 in all Makefiles to -O2. I'm using Red Hat 9 and it has been said that Red Hat does some funky stuff with their gcc so that may be causing it to segfault when compiled with -O6. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Does SIP work?
*This message was transferred with a trial version of CommuniGate(tm) Pro* Lack of documentation? Welcome to the bleeding edge... Enjoy.. J -Original Message- From: Tais M. Hansen [mailto:[EMAIL PROTECTED] Sent: Wednesday, 24 September 2003 10:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Does SIP work? *This message was transferred with a trial version of CommuniGate(tm) Pro* -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 September 2003 14:38, Andrew Kohlsmith wrote: Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. I use SIP to talk to Grandstream 100s every day, and also to the FWD network without issue. Are you trying to access SIP across NAT or other restrictive firewall or something? No firewall, no nat. Only lack of documentation for sip.conf... I got it to work now. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/cZQG2TEAILET3McRAu9MAJ9dGA6BVyTW/OBem/FZnzz1xBY9KACfbRge ILS+9IptUCB6HrsDDLVMrCA= =lRx5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
Title: Message I want to make it work and document it. So if somebody could send me some information I will be very pleased. Joe, what was your Cisco configuration ? - Original Message - From: Joseph Finley To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 11:06 AM Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz JozwiakSent: Wednesday, September 24, 2003 9:46 AMTo: ASTERISK USERSSubject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart
Re: [Asterisk-Users] Segmentation Fault on reload (gdb output included)
Sorry about that: bt gives the following output: #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 #1 0x40150c61 in malloc () from /lib/i686/libc.so.6 #2 0x40157dd0 in strdup () from /lib/i686/libc.so.6 #3 0x0805603b in cfg_process (tmp=0x80ea890, _tmpc=0x47a6a26c, _last=0x47a6a270, buf=0x65747865 Address 0x65747865 out of bounds, lineno=183, configfile=0x458d083d extensions.conf, includelevel=0) at config.c:57 #4 0x08055ac2 in __ast_load (configfile=0x458d083d extensions.conf, tmp=0x80ea890, _tmpc=0x47a6a26c, _last=0x47a6a270, includelevel=0) at config.c:731 #5 0x08055edd in ast_load (configfile=0x8 Address 0x8 out of bounds) at config.c:766 #6 0x458cf815 in pbx_load_module () at pbx_config.c:1543 #7 0x458ccdaf in reload () at pbx_config.c:1683 #8 0x08055372 in ast_module_reload () at loader.c:159 #9 0x0806b8ba in handle_reload (fd=44, argc=1, argv=0x47a6a5fc) at cli.c:105 #10 0x0806b6aa in ast_cli_command (fd=44, s=0x8 Address 0x8 out of bounds) at cli.c:1006 #11 0x08083c80 in netconsole (vconsole=0x80b9c28) at asterisk.c:192 #12 0x40021811 in pthread_start_thread () from /lib/i686/libpthread.so.0 I see extensions.conf mentioned a couple of times. Could this be caused by a configuration error? --- Steven Critchfield [EMAIL PROTECTED] wrote: On Wed, 2003-09-24 at 08:41, jerk face wrote: Ok, here is the real gdb output. This GDB was configured as i586-mandrake-linux-gnu... Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. ... ... ... Loaded symbols for /usr/lib/asterisk/modules/cdr_csv.so Reading symbols from /usr/lib/asterisk/modules/app_setcidnum.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_setcidnum.so #0 0x401519fc in mallopt () from /lib/i686/libc.so.6 Ok .. so what does this mean? Thank you in advance. This is where you type 'bt' and find out how it made it to that frame. --- Martin Pycko [EMAIL PROTECTED] wrote: actually gdb /usr/sbin/asterisk core.6044, sorry On Tue, 23 Sep 2003, jerk face wrote: I keep getting segmentation faults when I do a reload. Here are the core file outputs from gdb: (I have three of them and they produce the same output) (gdb) core core.6044 Core was generated by `asterisk'. Program terminated with signal 11, Segmentation fault. #0 0x401519fc in ?? () I have no idea what that means, but if somebody could point me in the right direction, that would be great. Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ 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 __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 errors?
I have the same but everything still seems to be working so I haven't worried about it.. maybe there has been an extention to the SIP protocol?? Later.. The following error messages were observed in /var/log/asterisk/messages: Sep 22 10:26:42 NOTICE[1133735216]: File chan_sip.c, Line 5099 (handle_request): Unknown SIP command 'PUBLISH' from '212.23.220.236' Sep 22 11:32:50 WARNING[1133735216]: File chan_sip.c, Line 4519 (handle_response ): Got 200 OK on REGISTER that isn't a register The phone was a Snom 200 running v2.01t code. The phone has been working fine. Are these misconfigurations on my part or something that I should be concerned about? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does SIP work?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 24 September 2003 16:36, Jamie Carl wrote: Lack of documentation? Welcome to the bleeding edge... I know, I just meant that pretty much everything else is either descriptive or described in sip.conf. Except the meaning of [xxx] entries. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/caz82TEAILET3McRAvD6AJ98K0jGtZLWTFh/OdjpGHxyAgz71gCfZIXI LbIVeURHr1OFLqQ/GEEZESU= =/k8y -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] THIS IS STRANGE
On Tue, 2003-09-23 at 09:44, Bartosz Jozwiak wrote: Right now it works great! Thanks so much. Could you tell me what is that: 'canreinvite=no' in sip.conf ? When SIP initiates the call, the INVITE message contains the information on where to send the media streams. * uses itself as the end-points of media streams when setting up the call, once the cal lhas been accepted, * sends another (re)INVITE message to the clients with the information to have the two clients send the media streams directly to each other. 'canreinvite=no' stops the sending of the (re)INVITEs once the call is established. From messages in the archives the ATA does not handle the (re)INVITE well. Also the thing to keep in mind is the call is being 'bounced' off of the asterisk server now. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Problems
Run the command festival Give it the command (SayText Would you like to play a game?) Does it say anything? If not, then there's a problem Festival. Type (Quit) to quit the festival app. On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote: I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say trying and then hangup... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Hello 1010 Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... Sep 24 23:05:33 WARNING[606231]: File app_festival.c, Line 410 (festival_exec): Festival WV command Any ideas? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) This message has been 'sanitized'. This means that potentially dangerous content has been rewritten or removed. The following log describes which actions were taken. Sanitizer (start=1064416439): Forcing message to be multipart/mixed, to facilitate logging. Writer (pos=843): Part (pos=917): Added 1 bytes of scratch space. Total modifications so far: 1 Added 1 bytes of scratch space. Anomy 0.0.0 : Sanitizer.pm $Id: Sanitizer.pm,v 1.79 2003/06/19 19:22:00 bre Exp $
RE: [Asterisk-Users] Check and restart script..
You can always use the safe_asterisk script...it's in the /usr/src directory. That's what I use. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 11:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Check and restart script.. Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running ps -aux |grep asterisk and then some kind of if to say that if the result is nothing then execute asterisk.. Problem with that theory is that the ps command will show up as well so i will have to work out a way to drop that.. Of course I may be missing a simpler or far better solution so thats why I am asking here first.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Check and restart script..
Could you not just add it to your /etc/inittab? -Original Message- From: WipeOut . [mailto:[EMAIL PROTECTED] Sent: 24 September 2003 16:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Check and restart script.. Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running ps -aux |grep asterisk and then some kind of if to say that if the result is nothing then execute asterisk.. Problem with that theory is that the ps command will show up as well so i will have to work out a way to drop that.. Of course I may be missing a simpler or far better solution so thats why I am asking here first.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Check and restart script..
Something like this can tell if asterisk is running. You can modify it as needed. Doesn't match the ps: if [ A`ps -e | grep asterisk | grep -v grep` = A ]; then echo echo It's not running echo else echo echo It's running echo fi Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 4:02 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Check and restart script.. Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running ps -aux |grep asterisk and then some kind of if to say that if the result is nothing then execute asterisk.. Problem with that theory is that the ps command will show up as well so i will have to work out a way to drop that.. Of course I may be missing a simpler or far better solution so thats why I am asking here first.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Check and restart script..
On Wed, 2003-09-24 at 10:01, WipeOut . wrote: Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running ps -aux |grep asterisk and then some kind of if to say that if the result is nothing then execute asterisk.. Problem with that theory is that the ps command will show up as well so i will have to work out a way to drop that.. I'm assuming you are doing something like ps axuwww|grep asterisk If you use a character class on the command line, the command line won't show up, but what your searching for will, ie. ps axuwww|grep [a]sterisk Of course I may be missing a simpler or far better solution so thats why I am asking here first.. Maybe you should look at init. From the init man page... When starting a new process, init first checks whether the file /etc/initscript exists. If it does, it uses this script to start the process. Each time a child terminates, init records the fact and the reason it died in /var/run/utmp and /var/log/wtmp, provided that these files exist. The first paragraph is important to make sure your modules are loaded at boot time, and the second line is important for debugging the reason why asterisk tripped up. Also init will make sure asterisk is running all the time you are in the appropriate runlevel you have defined. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival Problems
Did you Answer the call before calling Festival? Thorsten -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, September 24, 2003 11:13 To: [EMAIL PROTECTED] Run the command festival Give it the command (SayText Would you like to play a game?) Does it say anything? If not, then there's a problem Festival. Type (Quit) to quit the festival app. On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote: I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say trying and then hangup... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Hello 1010 Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... Sep 24 23:05:33 WARNING[606231]: File app_festival.c, Line 410 (festival_exec): Festival WV command Any ideas? Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Meridian Option 11 and asterisk
You can interface the two systems a variety of ways... the quick/easy route is a direct T1 from one switch to the other. You could do it via analog trunks too, but T1 signalling makes it so much smoother overall. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Farrell Sent: Wednesday, September 24, 2003 3:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11 and asterisk Has anyone ever interfaced a merdian option 11 and asterisk. Just wondering how you went about, it's for a small setup me only need between 4/6 channels, I was thinking about using some spare ISDN channels between the two. Has anyone seen an SIP option for the meridian? European Museum Of The Year 2002 The Chester Beatty Library http://www.cbl.ie/ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorised. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. ___ 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] Check and restart script..
3 ways: 1. in /etc/inittab : d1:23:respawn:/usr/sbin/asterisk -fv 2. use daemontools from DJB (this is what I use) 3 safe_asterisk (maybe is better this way) :-) WipeOut . wrote: Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running ps -aux |grep asterisk and then some kind of if to say that if the result is nothing then execute asterisk.. Problem with that theory is that the ps command will show up as well so i will have to work out a way to drop that.. Of course I may be missing a simpler or far better solution so thats why I am asking here first.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfert with dial plan
WipeOut . a crit: What problem are you having with tranfer on the GS phone? I have no clearly defined situation but I have experienced some lost calls during a call transfert with GS. Regards, Daniel Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
The configuration shouldn't be much different. Just replace port 1/0/0 with port 1/0:23, or whatever the voice port your PRI/T1/E1 happens to be. PLAR is Private Line Auto Ringdown. You pick up the port and it automatically dials. Think Batphone. In the configuration provided below it is configured so that anyone picking up the phone on the voice port (like an incoming phone call) will automatically dial , which then would connect to the IP address of Asterisk with H.323. -d Message: 4 Date: Wed, 24 Sep 2003 16:31:17 + (GMT) From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK Reply-To: [EMAIL PROTECTED] That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ 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 End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfert with dial plan
So it is working sometimes and then sometimes it doesnt.. I haven't had this problem.. I am using the .81 firmware and I use the following transfer process.. 1. Press the transfer button. 2. Dial the extension that I want to transfer to. 3. Press Redial button. (The Redial button has been renamed to Send on newer phones) Sorry I couldn't be of more help... WipeOut . a écrit: What problem are you having with tranfer on the GS phone? I have no clearly defined situation but I have experienced some lost calls during a call transfert with GS. Regards, Daniel Hello, As I have problems getting transfert call working with my grandstream SIP Phones, I woul like to know if it is possible to do it with a proper dial plan in exten.conf. I haven't found any information about that in the docs. Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200 errors?
WipeOut . schrieb: I have the same but everything still seems to be working so I haven't worried about it.. maybe there has been an extention to the SIP protocol?? ... me too. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Check and restart script..
Maybe you should look at init. From the init man page... When starting a new process, init first checks whether the file /etc/initscript exists. If it does, it uses this script to start the process. Each time a child terminates, init records the fact and the reason it died in /var/run/utmp and /var/log/wtmp, provided that these files exist. The first paragraph is important to make sure your modules are loaded at boot time, and the second line is important for debugging the reason why asterisk tripped up. Also init will make sure asterisk is running all the time you are in the appropriate runlevel you have defined. Hi Steven, Would you say init is better than safe_asterisk? or do they both cover the same bases.. Thanks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Problems
I have exactly the same symptoms with app_festival and I suspect that send_waveform_to_channel routine in app_festival.c doesn't work correctly. Festival works correctly since it sends wave file to asterisk, which saves it in cache. If I strip app_festival header in that file I can play it. The problem lies in playback of this wave to channel. Ant ideas? Best regards Borut Run the command festival Give it the command (SayText Would you like to play a game?) Does it say anything? If not, then there's a problem Festival. Type (Quit) to quit the festival app. On Wed, 2003-09-24 at 08:22, Bryan Nolen wrote: I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say trying and then hangup... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Hello 1010 Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... Sep 24 23:05:32 WARNING[606231]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... Sep 24 23:05:33 WARNING[606231]: File app_festival.c, Line 410 (festival_exec): Festival WV command ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfert with dial plan
WipeOut . a écrit: So it is working sometimes and then sometimes it doesnt.. I haven't had this problem.. I am using the .81 firmware and I use the following transfer process.. I am using this firmware too 1. Press the transfer button. 2. Dial the extension that I want to transfer to. 3. Press Redial button. (The Redial button has been renamed to Send on newer phones) and this process. Do you think that my problem may be related to chan_vpb driver? Daniel -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Problems
On Wed, 24 Sep 2003, Borut Senicar wrote: I have exactly the same symptoms with app_festival and I suspect that send_waveform_to_channel routine in app_festival.c doesn't work correctly. Festival works correctly since it sends wave file to asterisk, which saves it in cache. If I strip app_festival header in that file I can play it. The problem lies in playback of this wave to channel. Ant ideas? I didn't see an extensions.conf snippit that goes along with this, but I'm going to guess that the channel hasn't been Answered before the Festival app is being executed James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Check and restart script..
On Wed, 2003-09-24 at 11:57, WipeOut . wrote: Maybe you should look at init. From the init man page... When starting a new process, init first checks whether the file /etc/initscript exists. If it does, it uses this script to start the process. Each time a child terminates, init records the fact and the reason it died in /var/run/utmp and /var/log/wtmp, provided that these files exist. The first paragraph is important to make sure your modules are loaded at boot time, and the second line is important for debugging the reason why asterisk tripped up. Also init will make sure asterisk is running all the time you are in the appropriate runlevel you have defined. Hi Steven, Would you say init is better than safe_asterisk? or do they both cover the same bases.. From a quick look over the safe_asterisk script, it looks like a better option since it squirrels away your core files for you so you can do a better job of figuring out what went wrong. init was a top of the head type answer to try and point out how many tools already exist to fulfill the problem. I added it in the same manner I answered how to get ps to output the asterisk entries without the grep being included. Sometimes little tricks like that don't get passed from person to person unless you are sitting over the shoulder of someone who uses those shortcuts. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding a DELAY to an ADSI script
I was searching through the app_adsi.c file and found some events and functions that are not used in the sample ADSI scripts. One of these functions is DELAY. I can't get this to work. Has anybody got this to work? I'm trying to create a HangUp soft key using the following code: KEY Hangup IS Hang Up ONHOOK DELAY OFFHOOK ENDKEY The delay has no effect. Am I supposed to add arguments? Thank you for your time. __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
This is inbound FXO's pointed at the autoattendat on our * server. On Wed, 24 Sep 2003, Doug Dimick wrote: The configuration shouldn't be much different. Just replace port 1/0/0 with port 1/0:23, or whatever the voice port your PRI/T1/E1 happens to be. PLAR is Private Line Auto Ringdown. You pick up the port and it automatically dials. Think Batphone. In the configuration provided below it is configured so that anyone picking up the phone on the voice port (like an incoming phone call) will automatically dial , which then would connect to the IP address of Asterisk with H.323. -d Message: 4 Date: Wed, 24 Sep 2003 16:31:17 + (GMT) From: Sean Figgins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK Reply-To: [EMAIL PROTECTED] That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ 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 End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 2600 and ASTERISK
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml That covers the thridparty h323 stuff with * bkw On Wed, 24 Sep 2003, Sean Figgins wrote: That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ 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] VIA vs Intel
Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prebuilt Asterisk
Does anyone sell a preinstalled asterisk server ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfert with dial plan
Do you think that my problem may be related to chan_vpb driver? Daniel Dunno.. I am not that close to the internal workings of Asterisk.. Sorry.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
speaking of VIA - has anyone on the list looked at or used these ? http://www.mini-itx.com/store/default.asp?c=2currency=2 various collection of via based boards and cases and other goodies that go along with them. They are cheap enough they could work as either an asterisk server (diskless or with disk), or as phone platforms themselves. At 01:02 PM 9/24/2003 -0500, you wrote: On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
On Wed, 2003-09-24 at 10:41, Mike Hjorleifsson wrote:
Has anyone successfully run asterisk with a VIA processor ?
I have tried unsucessfully, do I have to run make with any specific switches
?
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I once used to fight that problem... The VIA would aspire to be a i686,
but it is actually a i586 from a compiler standpoint. So with that, you
need to make a change to you /usr/src/asterisk/Makefile and change the
following:
#ifeq (${OSARCH},Linux)
#PROC=$(shell uname -m)
#endif
# Pentium Pro Optimize
#PROC=i686
#
Pentium VIA processors optimize
PROC=i586
Make sure that you comment out the ifeq (${OSARCH},Linux) or else when
the Makefile is running, it will reset the PROC value from i586 back to
i686.
K.
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Re: [Asterisk-Users] Prebuilt Asterisk
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934130944.htm bkw On Wed, 24 Sep 2003, Mike Hjorleifsson wrote: Does anyone sell a preinstalled asterisk server ? ___ 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] Prebuilt Asterisk
Mike Hjorleifsson wrote: Does anyone sell a preinstalled asterisk server ? I believe TelAppliant.co.uk will sell you a system called mypbx, which is basically just that. -- Alastair Maw MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization
Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 2600 and ASTERISK
This is my configuration of my cisco router and still it does not want to work :( Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname asterisk ! aaa new-model aaa authentication login default local enable secret 5 $1$bJzJ$bjJ.hc0TbiopbjjMUnyhg/ ! username admin password 7 07002C494908 ! ! ! ! ip subnet-zero ip name-server 66.178.37.211 ! ! ! ! voice-port 1/0/0 ! voice-port 1/0/1 ! voice-port 1/1/0 ! voice-port 1/1/1 connection plar ! ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ! interface Serial0/0 no ip address no ip directed-broadcast shutdown ! interface Ethernet0/1 no ip address no ip directed-broadcast shutdown half-duplex ! ip classless ip route 0.0.0.0 0.0.0.0 66.178.36.4 no ip http server ! ! line con 0 transport input none line aux 0 line vty 0 4 ! no scheduler allocate end - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 2:25 PM Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml That covers the thridparty h323 stuff with * bkw On Wed, 24 Sep 2003, Sean Figgins wrote: That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
On Wed, 2003-09-24 at 13:13, Jon Pounder wrote: speaking of VIA - has anyone on the list looked at or used these ? http://www.mini-itx.com/store/default.asp?c=2currency=2 various collection of via based boards and cases and other goodies that go along with them. They are cheap enough they could work as either an asterisk server (diskless or with disk), or as phone platforms themselves. I was just looking at them since someone has built a mini distro to make one of these devices into a MythTV front end. I could see spending $200 per TV in my house to front them with these little boxes and then fill a couple machines up in a rack in the basement taping shows for the family. But, this is the wrong list to finish talking about this subject. As a phone platform, it may be overkill, but I bet it could drive a TDM400P card and be able to handle GSM compression. The question then again is if it is worth the cost for basically a 4 port asterisk based device like the ATA186? At 01:02 PM 9/24/2003 -0500, you wrote: On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2Lang=ENG I bought it via http://mini-itx.com/ quote who=Steven Critchfield On Wed, 2003-09-24 at 12:41, Mike Hjorleifsson wrote: Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ? Yes, look for comments about a 586 flag since the via chips aren't fully PII or above compatible. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Undocumented variables in chan_sip.c
I haven't actually read the code involved, but my *guess* would be that setting srvlookup to yes means that the NAPTR/SRV lookup procedure described in RFC 3263 is used to turn SIP hostnames into an IP addresses. It's also possible that it means that Asterisk will use the older, deprecated procedure from RFC 2543 (which uses just SRV records without looking NAPTR records first). /a -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Monday, September 22, 2003 15:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Undocumented variables in chan_sip.c Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify: * srvlookup=yes|no * pedantic * canreinvite=update|yes --update seems new Being curious, especially for srvlookup functionality... /O ___ 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] RedHat 9.0 and 100 percent CPU utilization
i run some system with * rh 9.0 be sure to have latest updates (install them after installing redhat, and before installing *) and check to have mpg123 installed (you must get it on the mpg123 website), since redhat has a mpg321 replacement that won't work with * matteo. Il mer, 2003-09-24 alle 20:23, James Ray ha scritto: Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
On Wed, 24 Sep 2003, Steven Critchfield wrote: As a phone platform, it may be overkill, but I bet it could drive a TDM400P card and be able to handle GSM compression. The question then again is if it is worth the cost for basically a 4 port asterisk based device like the ATA186? I have a mini-itx board (800mhz) and case (I can look up the part number if anyone is interested), but the older TDM400P has the sound problems with the power supply in there. I've been meaning to contact digium to swap the card out and try a new rev in there but I haven't had the chance. For now I have a T100P working in there great. I originally wanted to get the TDM400P working in there because its a great demo system to show people just what it can do. People are very impressed when you can walk in, plug a box in the ethernet (assuming dhcp), plug a phone into the back, and start making calls via IAX. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prebuilt Asterisk
Contact Sean Robertson at NETXUSA for prebuilt system. [EMAIL PROTECTED] 1-864-271-9868 1-800-292-0728 - Original Message - From: Mike Hjorleifsson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 1:43 PM Subject: [Asterisk-Users] Prebuilt Asterisk Does anyone sell a preinstalled asterisk server ? ___ 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] No ring tone while dialing out with AVM PCI2.0
Hi, I use an AVM FRITZ PCI 2.0 to dial out but although it works OK and places the call there is no ring or busy tone. Has someone figured out this problem? Thanks __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2Lang=ENG I bought it via http://mini-itx.com/ How many concurrent calls have you run on this MB?? What codecs are you using for your phones? (assuming IP phones) Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0
Hi Jim, get chan_capi from www.junghanns.net/asterisk/ install the capi drivers for your card from ftp.avm.de/cardware best regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Mit, 2003-09-24 um 21.17 schrieb Jim Paraschou: Hi, I use an AVM FRITZ PCI 2.0 to dial out but although it works OK and places the call there is no ring or busy tone. Has someone figured out this problem? Thanks __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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
AW: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0
Hi Jim, i had the same probs, and it seems to be bug/feature of i4l. I can not find anything in the code that would bring these messages to the top of ttyI:-( Or is there somebody who knows it better ??? ;-) Regards, Thomas. -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Jim Paraschou Gesendet: Mittwoch, 24. September 2003 21:18 An: [EMAIL PROTECTED] Betreff: [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0 Hi, I use an AVM FRITZ PCI 2.0 to dial out but although it works OK and places the call there is no ring or busy tone. Has someone figured out this problem? Thanks __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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
Re: [Asterisk-Users] VIA vs Intel
quote who=WipeOut . I am running Asterisk on one of these (T100P taking the single PCI slot.) (EPIA M1 Mini-ITX Motherboard) http://www.hushtechnologies.com/default.asp?pageID=2Lang=ENG I bought it via http://mini-itx.com/ How many concurrent calls have you run on this MB?? What codecs are you using for your phones? (assuming IP phones) Later.. -- Don't know yet. I just got my GrandStream phones yesterday, quantity four. Got one configured and dialed into VoiceMail2() just fine. (ULaw) I have it hooked up to a channelbank, but still need to get an RJ21 cable to connect to my breakout box. (CAC AB1) I have this setup sitting on my desk at work. I will need to bring it home to connect to analog trunk lines. -- END OF LINE ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Festival Problems
I have exactly the same symptoms with app_festival and I suspect that send_waveform_to_channel routine in app_festival.c doesn't work correctly. Festival works correctly since it sends wave file to asterisk, which saves it in cache. If I strip app_festival header in that file I can play it. The problem lies in playback of this wave to channel. Ant ideas? I didn't see an extensions.conf snippit that goes along with this, but I'm going to guess that the channel hasn't been Answered before the Festival app is being executed It is answered but * just drops the connection when reached Festival application in dialplan: [home-trusted] exten = 1000,1,answer exten = 1000,2,SayDigits(123) exten = 1000,3,Festival(Connect to Festival) exten = 1000,4,Wait(5) exten = 1000,5,Festival(send the argument) exten = 1000,6,Hangup Trace from console: -- Executing Answer(SIP/bsenicar-850b, ) in new stack -- Executing SayDigits(SIP/bsenicar-850b, 123) in new stack -- Playing 'digits/1' -- Playing 'digits/2' -- Playing 'digits/3' -- Executing Festival(SIP/bsenicar-850b, Connect to Festival) in new stack == Parsing '/etc/asterisk/festival.conf': Found WARNING[147466]: File app_festival.c, Line 304 (festival_exec): Text passed to festival server : Connect to Festival WARNING[147466]: File app_festival.c, Line 353 (festival_exec): line length : 19 WARNING[147466]: File app_festival.c, Line 357 (festival_exec): Seek position : 23 WARNING[147466]: File app_festival.c, Line 381 (festival_exec): Passing text to festival... WARNING[147466]: File app_festival.c, Line 390 (festival_exec): Writing result to cache... WARNING[147466]: File app_festival.c, Line 400 (festival_exec): Passing data to channel... == Spawn extension (home-trusted, 1000, 3) exited non-zero on 'SIP/bsenicar-850b' Somebody suggested that enclosing arguments with single quotes will help, but in case result was same. Best regards Borut ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More onCallprogress
Here is some more stuff to add to the confusion about the callprogress option. I currently have my * system operating with a T100P talking to an ADTRAN TSU600 channel bank with 8 FXO ports connecting to the outside world and Grandstream SIP phones as handset extensions. At first I naively set callprogress=yes in zapata.conf. The results were typical of what many people have reported in the lists: I could receive incoming calls (to the SIP hardware) with no problem but outgoing calls failed. When callprogress=no was used, everything was fine. Given this, I assumed that * cannot tell that the remote (analog) line had been answered, so the call was never bridged. I concluded that the ADTRAN TSU would have to detect answered status, and that therefore this was the expected behavior: callprogress should only be able to operate on FXO's that are plugged directly into the * box. What's strange is that as long as the remote phone does not pick up, the * console indicates ringing on the zap interface (i.e., Zap/1-1 Ringing repeats). However, when the remote phone picks up, * stops announcing that it is detecting ringing and waits (forever, or at least until the remote line is hung up). In other words, either the ADTRAN had signalled that ringing had stopped, or * was listening and noticed that ringing had stopped. However, even though the remote phone is picked up, I still hear (the locally generated) ringback on the SIP phone, indicating that * had not acknowleged the remote phone having been picked up, even though it apparently knows - it has after all stopped repeatedly displaying the ringing notification. The questions are these: 1) If callprogress is in fact detecting ringing, and then the cessation of ringing through the ADTRAN, then why doesn't it bridge the call when the ringing stops? 2) Why bother at all with callprogress. If bridging an analog call when it's dialed (i.e., callprogress=no) always succeeds, then why wait? The only advantage I can see is that it may save some wasted bandwidth, which, while not important in my case, could be very important to people with large call volumes. But then, those people would probably not be using analog lines anyway so the problem goes away. Can someone clarify? 3) Finally, if I disable busydetect, what are the consequences? Does it simply mean that * won't be able to easily detect when the remote end of the analog line hangs up? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs
You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens. -d Message: 10 From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 2600 and ASTERISK Date: Wed, 24 Sep 2003 15:29:22 -0300 Organization: Cq-Link, Parbo Net Reply-To: [EMAIL PROTECTED] This is my configuration of my cisco router and still it does not want to work :( Current configuration: ! version 12.0 service timestamps debug uptime service timestamps log uptime service password-encryption ! hostname asterisk ! aaa new-model aaa authentication login default local enable secret 5 $1$bJzJ$bjJ.hc0TbiopbjjMUnyhg/ ! username admin password 7 07002C494908 ! ! ! ! ip subnet-zero ip name-server 66.178.37.211 ! ! ! ! voice-port 1/0/0 ! voice-port 1/0/1 ! voice-port 1/1/0 ! voice-port 1/1/1 connection plar ! ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:66.178.37.169 ! ! interface Ethernet0/0 ip address 66.178.37.169 255.255.254.0 no ip directed-broadcast half-duplex ! interface Serial0/0 no ip address no ip directed-broadcast shutdown ! interface Ethernet0/1 no ip address no ip directed-broadcast shutdown half-duplex ! ip classless ip route 0.0.0.0 0.0.0.0 66.178.36.4 no ip http server ! ! line con 0 transport input none line aux 0 line vty 0 4 ! no scheduler allocate end - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 2:25 PM Subject: RE: [Asterisk-Users] Cisco 2600 and ASTERISK http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080093f62.shtml That covers the thridparty h323 stuff with * bkw On Wed, 24 Sep 2003, Sean Figgins wrote: That is about what I have been seing for help. Has anyone any clue what to di with a 2600 that has a T1 adapter on a high-density high-density voice port adapter? BTW... Because I am lazy, what does plar do? -Sean On Wed, 24 Sep 2003, Brian West wrote: This is simple to do.. voice-port 1/0/0 connection plar ! voice-port 1/0/1 connection plar ! dial-peer voice 1000 voip max-conn 4 destination-pattern req-qos guaranteed-delay codec g711ulaw ip precedence 5 no vad session target ipv4:x.x.x.x ! in h323.conf set the context=blah [blah] exten = ,1,Goto(s,1) Done... its really that simple. I have this working with a 2600 and a 1750. bkw On Wed, 24 Sep 2003, Joseph Finley wrote: I too would like to see it. I've tried many times with the help of a few and never got it to work. It always results in a fast busy. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bartosz Jozwiak Sent: Wednesday, September 24, 2003 9:46 AM To: ASTERISK USERS Subject: [Asterisk-Users] Cisco 2600 and ASTERISK Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE:# [Asterisk-Users] No ring tone while dialing out with AVM PCI2.0
Thank you. Is there any documenantation available abount installing and configuring chan_capi? __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi accountcode.. (repost)
Hi, All my inbound calls have a blank account code in the CDR.. Where or what is the correct way to set the accountcode= setting when using chan_capi channels? Do I do it in capi.conf or extensions.conf with a setvar? or some other way.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dlink DG-104S (chan_mgcp) and configuration w/Asterisk
I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: [EMAIL PROTECTED]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) -- don't have a manual for it... In my mgcp.conf I have: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [dlinkgw] host = 172.16.1.42 context = default line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 But I'm getting errors spew'ing on the * console: NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/1') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/3') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/2') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/4') does not exist Any ideas or does someone have a prefer config that works... Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL:http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dlink DG-104S (chan_mgcp) and configuration w/Asterisk
This is what I have in my mgcp.conf [dlink] threewaycalling=yes transfer=yes callwaiting=yes callwaitingcallerid=yes host=dynamic context=international nat=yes ;dtmf=inband disallow=all allow=g711 allow=ulaw callerid = Andrew Joakimsen 321 line = aaln/1 callerid = Andrew Joakimsen 322 line = aaln/2 callerid = Andrew Joakimsen 323 line = aaln/3 callerid = Andrew Joakimsen 324 line = aaln/4 I would at least change the host=dynamic (or host=ip of dlink) In the dlink I have the following set: Notify Entity: [EMAIL PROTECTED]:2427 RGW Name: dlink I am not sure if it will work if the * server is on a public IP and the dlink behind a NAT. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 5:01 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dlink DG-104S (chan_mgcp) and configuration w/Asterisk I have a DG-104S (which I reset to factory settings, it's DHCP'ing off my network, plugged into the WAN port). The system comes up, and I through the web browser set under Call Agent IP Address to: Notify Entry: [EMAIL PROTECTED]:2427 (192.168.1.1 is the * server) I have RGW Name: and DNS IP addres the DNS IP of the MGCP box and DNS State disabled (not sure what to set it to) -- don't have a manual for it... In my mgcp.conf I have: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [dlinkgw] host = 172.16.1.42 context = default line = aaln/1 line = aaln/2 line = aaln/3 line = aaln/4 But I'm getting errors spew'ing on the * console: NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/1') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/3') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/2') does not exist NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '172.16.1.42' (and thus its endpoint 'aaln/4') does not exist Any ideas or does someone have a prefer config that works... Thanks, Lenny --- Lenny Tropiano E-mail: [EMAIL PROTECTED] Partner, Networking Specialist Pager: [EMAIL PROTECTED] VoIPing, LLCURL: http://www.voiping.com/ PO Box 867, Cedar Park, TX 78630-0867 Mobile: 512-698-VOIP [8647] ___ 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] RedHat 9.0 and 100 percent CPU utilization
Do you have any dtmfmode=inband in you sip.conf? Regards, Gus - Original Message - From: Scott Stingel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 24, 2003 5:13 PM Subject: RE: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization Lot's of people on here use Redhat 9.0 - don't worry! The 100% utilisation sounds wrong, assuming that asterisk is actually handling any calls though. Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Ray Sent: Wednesday, September 24, 2003 7:23 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RedHat 9.0 and 100 percent CPU utilization Please, don't hate me because I use Redhat. I am aware that I am asking for problems in running Asterisk on Redhat. I recently aquired a nifty server, moved my digium cards, and installed asterisk. I noticed that one of the four processors was being used at 100% and nothing was working. I tracked CPU utilization back to the Asterisk process. Please, help. James __ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.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 mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Very valuable help. It is now working like a champ. This is a solution with SIP--NAT---Internet---Asterisk. No problems here. What I would like to do next is to move Asterisk behind a NAT as follows SIP---NAT---Internet---NAT---Asterisk do I need a STUN server? is there a chance this could work? The Google results seems to indicate that I will get an ulcer attempting this step. is that true? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration Try adding nat=yes to your config.. Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no.. Finally set dtmfmode=info for the GS phones.. Later.. Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf --- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a register statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VIA vs Intel
James Golovich wrote: On Wed, 24 Sep 2003, Steven Critchfield wrote: As a phone platform, it may be overkill, but I bet it could drive a TDM400P card and be able to handle GSM compression. The question then again is if it is worth the cost for basically a 4 port asterisk based device like the ATA186? I have a mini-itx board (800mhz) and case (I can look up the part number if anyone is interested), but the older TDM400P has the sound problems with the power supply in there. I've been meaning to contact digium to swap the card out and try a new rev in there but I haven't had the chance. For now I have a T100P working in there great. I originally wanted to get the TDM400P working in there because its a great demo system to show people just what it can do. People are very impressed when you can walk in, plug a box in the ethernet (assuming dhcp), plug a phone into the back, and start making calls via IAX. James Got a similar setup, using a Mini-itx C3 800MHz in a Casetronix 2677 case. Like you, I have noise problems with the TDM400. My solution: use a Dlink 104S (US$80 off ebay). Now my box has 1 XP101 in the PCI slot and a cross cable to the Dlink. A really neat demo set that actually fits in a weekend bag. My only gripe with the C3, its FPU is slo (IIRC it's 1/2 clock speed). Right now, I'm thinking of getting the Nehemiah (full speed FPU). Have you taken a look at http://www.caseoutlet.com/NWPc/C137/c137_barebone.html? It's about US$400 for the 2 PCI slot version. Leo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible - Original Message - From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration Very valuable help. It is now working like a champ. This is a solution with SIP--NAT---Internet---Asterisk. No problems here. What I would like to do next is to move Asterisk behind a NAT as follows SIP---NAT---Internet---NAT---Asterisk do I need a STUN server? is there a chance this could work? The Google results seems to indicate that I will get an ulcer attempting this step. is that true? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration Try adding nat=yes to your config.. Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no.. Finally set dtmfmode=info for the GS phones.. Later.. Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf --- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a register statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help asterisk call waiting X100P - MGCP ata 186
I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk - ATA 186 MGCP -- analog phone. How do I answer the call waiting beep.. Thanks, I appreciate any help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo for 15 seconds
Hello, I am running asterisk with two X100P cards using a cisco ata 186 "MGCP" for phone connections. For the first 15 seconds of a call I get echo on the ata 186 side only. I assume after that the echo canceller kicks in but is there any way to make it happen faster? I have read some stuff about this on the threads but didn't find the answer. I appreciate any help very much. Thanks Chad
[Asterisk-Users] (no subject)
Dear All, I am going to deploy a VOIP network here in Canada with nodes all over town. This is for long distance services and hence would need a good reliable solution. I have looked into * and am very interested in it with all the value-added features as well as its capability to do H323 and SIP. I understand that a good portion of VOIP operators in the industy is converting to SIP but there are still more H323 VOIP operators out there. That is one reason why I am interested in this solution as it can do both. Most of my customers and carriers are still on H323 and hence I would need to make sure that the * is able to talk with most H323 gateways out there in the market, such as cisco and quintum. There are two things I would like your comments on to make sure that the system will serve my purpose. I need my carrier customers to be able to send calls to my * via H323 VOIP from most H323 gateways and the * to pass through and route the calls to appropriate carriers with cisco or other gateways via VOIP, the process will be H323 mostly and some SIP, working with G723 and / or G729 which are what most VOIP operators are using. In this situation, the inbound and outbound are both via VOIP, either with H323 or SIP with both G723 and / or G729 codec. Is the application, is * compatible with most gateways out there? Another situation is carrier customers will send calls to me to be terminated on the TDM circuits OR I originate calls from the TDM side to be terminated on any gateways on the carrier side. In such, there will be a encoding / decoding process. Would I be able to send calls to most other gateways from my TDM circuits via VOIP H323 or SIP on g729 / G723 codec? Likewise for customers calling into my *, would the * be able to decode G729 and G723 to pass calls to the TDM side? I would love to get some feedbacks and advices (which is greatly appreciated) from you all who have had experiences in doing this before, thanks amillion. Tommy Chan
[Asterisk-Users] Removal of anti-spam responder
As others have said, everything I post to asterisk users, I get this anti-spam HTML saying please authenicate. This email indicates it's on behalf of [EMAIL PROTECTED] Can we please remove this person from the list. thanks, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Wednesday, September 24, 2003 7:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible - Original Message - From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration Very valuable help. It is now working like a champ. This is a solution with SIP--NAT---Internet---Asterisk. No problems here. What I would like to do next is to move Asterisk behind a NAT as follows SIP---NAT---Internet---NAT---Asterisk do I need a STUN server? is there a chance this could work? The Google results seems to indicate that I will get an ulcer attempting this step. is that true? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration Try adding nat=yes to your config.. Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no.. Finally set dtmfmode=info for the GS phones.. Later.. Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf --- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a register statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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
