[asterisk-users] IAX Trunking Only ONE-WAY
Hello, i have been reading the archives and cant find the answer to my
question. Here is the scenario,
SIP dallas gateway -- IAX2 g729 codec -- mex gw
So i send all my sip traffic on iax2 to mexico using g729 and trunk
mode, but the trunk debug shows only mexico doing trunking, in mexico
i get RX BW 40% higher than TX BW.
on dallas server when i get:
IAX2 Trunk Debug Requested
Beginning trunk processing. Trunk queue ceiling is 128000 bytes per
host
- Trunk peer (200.xxx.xxx.190:4569) has 0 call chunks in transit, 0
bytes backloged and has hit a high water mark of 0 bytes
Ending trunk processing with 1 peers and 0 call chunks processed
on mexico server i get:
Beginning trunk processing. Trunk queue ceiling is 128000 bytes per host
- Trunk peer (216.xxx.xxx.254:4569) has 19 call chunks in transit, 0
bytes backloged and has hit a high water mark of 6400 bytes
Ending trunk processing with 1 peers and 19 call chunks processed
So my bw monitor tool shows this on mexico
IfaceRX(KB/sec) TX(KB/sec) Total(KB/sec)
lo0.0000.000 0.000
eth0 62.023 27.814 89.837
sit00.0000.000 0.000
Total 62.023 27.814 89.837
My configuration on Dallas (Sending Calls)
iax.conf
[general]
bindaddr=216.xxx.xxx.254
bindport=4569
bandwidth=low
echocancel=no
jitterbuffer=no
tos=lowdelay
disallow=all
allow=g729
[mtygw]
type=friend
trunk=yes
context=mexico-out
username=mtygw
secret=***
notransfer=yes
host=200.xxx.xxx.190
disallow=all
allow=g729
extensions.conf
MTYGW=IAX2/dallas:[EMAIL PROTECTED]
exten = _0115281.,1,Dial(${MTYGW}/${EXTEN:7},31)
My configuration on Mexico (Reciving Calls)
iax.conf
[general]
bindaddr=200.xxx.xxx.190
bindport=4569
bandwidth=low
echocancel=no
disallow=all
allow=g729
allow=gsm
jitterbuffer=no
tos=lowdelay
[dallas]
type=friend
trunk=yes
context=mexico-out
username=dallas
secret=*
host=216.xxx.xxx.254
notransfer=yes
disallow=all
allow=g729
mexico only gets calls from dallas so theres no need to show you the
extensions.conf setup. Any ideas what can it be wrong that i get only
trunking on one side? thank you guys
p.s. im getting the timers on dallas from ztdummy im using kernel
2.6.12-10-686-smp and im getting the timer on mexico from the digium
card.
Miguel Cavazos
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[Asterisk-Users] Asterisk Realtime 2 Servers calling each other
Hi guys I have a question, im trying asterisk realtime in 2 servers. Im trying to make calls from one server to another, example I call a sip registered in sip server 1 with a phone register in sip server2 and both using the same database and family both use canreinvite=yes but still cant make the calls any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Realtime regseconds update
Hi guys, im using realtime and I want to show registered users or online users on a webpage and offline users. Im taking regseconds field to make this happend If regseconds value is 0 then user appers offline, it regseconds is something else then its online, but sometimes this works and sometimes it does not. Im using the following options rtcachefriends=yes rtnoupdate=yes rtautoclear=yes anyone has any idea? im using 1.2.0beta1, im not sure if its updating this field, i have on also set in my sip.conf file defaultexpirey=300 maxexpirey=300 Also my atas, are set with this value, so it should expire in 300 seconds but sometimes this doesnt occure. Miguel ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 problem loading
hi guys im getting this error when trying to load chan_h323 on my local box Mar 16 17:19:27 WARNING[2278]: libh323_linux_x86_r.so.1.12.2: cannot open shared object file: No such file or directory Mar 16 17:18:36 WARNING[2265]: Loading module chan_h323.so failed! any ideas? everything compiled well -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 in Bolivia
What version are you using for chan_unicall? On 25/01/2005, at 1:57 PM, [EMAIL PROTECTED] wrote: Hi I made some tests with new MFC/R2 an unicall support for asterisk and now have dialing out problem using UniCall / R2. This is the error report in cli UC channel 30 protocol error. Cause 32772 I hope this helps. Thanks in advance, Jorge PD: de conf files zaptel.conf span=1,1,1,cas,hdb3 cas=1-15:1010 cas=17-31:1010protocolclass=mfcr2 unicall.conf protocolvariant=bo,20,4 protocolend=cpe group = 3 channel = 1-15 ;skip time slot 16 channel=17-31 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Debian woody stable, nothing special most of the trouble are paths On 13/01/2005, at 1:26 AM, Sam Njenga wrote: Hi Am setting up * with R2/MfC support but am 90% done. I seem to be missing something in my setup. Can you tell me what Linux distribution and the packages you have used to complete your setup to a working level ? /Sam - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, January 13, 2005 1:22 AM Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
any feedback would be awsome, the idea is to fill in the 30 channels of the E1 all at the same time and see how stable it can be On 13/01/2005, at 8:28 AM, Don Dawson wrote: I have an asterisk system down here in Oaxaca. I don't know anyone there to call but I can call some hotels in the area for possible reservations and perhaps ticket information for the theater. - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, January 12, 2005 4:22 PM Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
On 13/01/2005, at 9:35 AM, Miguel Cavazos wrote:
Really weird calls are still getting in and i just called the same
number as you did. I will investigate.
here is the context on extensions.conf
[guest]
exten = _,1,Dial(Unicall/g1/${EXTEN},90,Tt)
On 13/01/2005, at 9:22 AM, Gary Carr wrote:
I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d,
IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
-- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read:
Call rejected
by 200.53.121.233: No such context/extension
-- Hungup 'IAX2/200.53.121.233:4569/4'
== No one is available to answer at this time
Regards,
Gary
- Original Message - From: Miguel Cavazos
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 10:13 AM
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
any feedback would be awsome, the idea is to fill in the 30 channels
of the E1 all at the same time and see how stable it can be
On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone
there to
call but I can call some hotels
in the area for possible reservations and perhaps ticket
information for the
theater.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 4:22 PM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
Hi guys, I have one E1 with 30 channels in Mexico City, I guess
that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can see how stable it can be. Im
using R2/MFC with chan_unicall the patch that Steve Underwood
wrote.
I will let anyone make FREE LOCAL calls to Mexico City till
saturday or
maybe until monday to see how stable this can be with REAL
traffic. Add
this to your extensions.conf only gsm as a codec is going to be
permitted.
exten =
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
--
Saludos,
Miguel Cavazos
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Saludos,
Miguel Cavazos
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--
Saludos,
Miguel Cavazos
--
Saludos,
Miguel Cavazos
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Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
For those who doesnt have an asterisk setup or cant make it work you
can use any iaxsoftphone and use the user guest with no password using
codec gsm and start dialing as if you are in mexico city. We need to
have alot of calls going! The ip for the server is 200.53.121.233
On 13/01/2005, at 11:05 AM, Greg Blakely wrote:
Works for me, too. But I found that the Benito Juarez International
airport was reachable by 9-011-52-5-571-3600.
To get this from my PBX-like setup, I have the following in
extensions.conf:
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt)
and the following in iax.conf
disallow=all
allow=GSM
allow=ULAW
allow=ALAW
allow=G726
allow=ILBC
allow=LPC10
allow=SPEEX
(Obviously, anything below allow=GSM isn't necessary for this
particular connection.)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Don Dawson
Sent: Thursday, January 13, 2005 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test
chan_unicall
I changed to line to :
exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt
and it works fine.
On 13/01/2005, at 9:22 AM, Gary Carr wrote:
I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d,
IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
-- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339
socket_read:
Call rejected
by 200.53.121.233: No such context/extension
-- Hungup 'IAX2/200.53.121.233:4569/4'
== No one is available to answer at this time
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Saludos,
Miguel Cavazos
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Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Thanx but that is consider in Mexico bypass and its illegal, second we are just doing a test with real traffic to get feedback of any weird thing going on. Testing Chan_unicall stability is our goal. If you can send alot of traffic while we are doing test i would thank you for that. Till now we have only got 4 channels at most busy and we need to see if it will handle a full E1 to test then with 2,3 and 4 E1's On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote: I tried to contact you off list, but your system rejected my e-mail, I was wondering ig you planed on selling minutes for routes into Mexico once you where done testing, if so, could you please contact me off list with your rates for Mexico City, or anyplace else in Mexico you service, thank you. Nathan Goodwin Diamondleaf LLC Miguel Cavazos wrote: Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Yes, thats why i will do it for a very short time to do testing with real traffic. On 13/01/2005, at 4:03 PM, Nathan Goodwin wrote: Wouldn't that make routing free calls illegal as well, your still bypassing? Miguel Cavazos wrote: Thanx but that is consider in Mexico bypass and its illegal, second we are just doing a test with real traffic to get feedback of any weird thing going on. Testing Chan_unicall stability is our goal. If you can send alot of traffic while we are doing test i would thank you for that. Till now we have only got 4 channels at most busy and we need to see if it will handle a full E1 to test then with 2,3 and 4 E1's On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote: I tried to contact you off list, but your system rejected my e-mail, I was wondering ig you planed on selling minutes for routes into Mexico once you where done testing, if so, could you please contact me off list with your rates for Mexico City, or anyplace else in Mexico you service, thank you. Nathan Goodwin Diamondleaf LLC Miguel Cavazos wrote: Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
I do have a WISIP and it doesnt give me any problems im all day long on the street using it. You cant talk of a phone you havent even touch Miguel On Fri, 2004-04-23 at 10:33, Andrew Kohlsmith wrote: why not wisip? its size its like a regular cellphone and it uses wifi Because it sucks ass? Check the archives for some very valid gripes about the device. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smallest phone
why not wisip? its size its like a regular cellphone and it uses wifi Miguel Cavazos On Fri, 2004-04-23 at 08:00, Chris Hirsch wrote: Tim Sailer wrote: Folks, I'm looking for a SIP or IAX phone for field techs to take with them when out on service calls. The regular desktop phones are just way too big. Is there anything like the size of a full-sized cell phone? Or smaller, not I doubt that... If a softphone is acceptable what about something like http://www.kauss.org/Stephan/ziaxphone/ Can't get much smaller than that :-) -- The older you get, the better you realize you were. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] About IAX channels
IAX was removed on newer versions replace it with IAX2 just make sure to change on your extensions.conf IAX/user:[EMAIL PROTECTED]/201 to IAX2/user:[EMAIL PROTECTED]/201 Good luck Miguel On Wed, 2004-04-21 at 03:44, Jan Madsen wrote: I have been running af Asterisk server Version 0.7.2 for a while now But I I wanted to upgrade my version to the new 0.9.0 or the CVS 1.0 Stable. But when I install one of the new asterisk servers I having lots of troubles with the IAX connection between my servers. When I start the 0.7.2 asterisk server it shows me something lige this == Parsing '/etc/asterisk/iax.conf': Found == Using TOS bits 16 == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver) == Registered channel type 'IAX' (Inter Asterisk eXchange Drver) == IAX Ready and Listening on 0.0.0.0 port 5036 As you can see the Asterisk 0.7.2 registering the Inter Asterisk eXchange Drver But when I start the other 2 versions these lines don't appear. Instead when I try to make a call over the IAX lines the 0.9.0 and 1.0 versions print in the console -- Executing Dial(SIP/302-c0c3, IAX/user:[EMAIL PROTECTED]/201) in new stack Apr 21 09:35:38 WARNING[-1210897488]: channel.c:1676 ast_request: No channel type registered for 'IAX' Apr 21 09:35:38 NOTICE[-1210897488]: app_dial.c:536 dial_exec: Unable to create channel of type 'IAX' == Everyone is busy at this time -- Executing Congestion(SIP/302-c0c3, ) in new stack == Spawn extension (default, 201, 2) exited non-zero on 'SIP/302-c0c3' -- Registered to '192.168.24.100', who sees us as 192.168.24.101:4569 I don't know how to get these IAX lines to work on the 1.0 and 0.9.0 versions do someone know how to do this Thanks for any response I will get Best regards Jan Madsen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 oh323 g729 please help !
update your crappy hardware :)?? atleast with sip you will be able to allow both codecs. Miguel On Mon, 2004-04-19 at 07:51, Serge wrote: Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): - asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable all other codec ! ( bug ? ) , but I need minimum 2 - g711 and g729. h323 -- all work ok , but only for new phones ! like cisco ATA .., with this driver old phones don't may speak with asterisk ! So, and last bug.. when I enable 2 codec in both version, I need DTMF inbound ( for g711 ) , but all time error, due g729 enabled. Can I set codec by destination? ( like SIP ) I try use 2 cnannels at the same time, but asterisk down with segmentation fault... Thanks, Serge. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anyone has a working * with E1 in Mexico E1 R2 modified?
Yes it works with normal digium hardware Miguel On Thu, 2004-04-01 at 08:51, Otto Krumm wrote: I was wondering if anyone has setup an * connected to E1 in Mexico?, what card would you recomend and do you have some info, examples or everythig else... or for instance this setup works? Thanks in advance Greetings Otto Krumm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 variants and Asterisk
si funciona con el A y B Miguel Cavazos On Thu, 2004-03-25 at 22:47, Carlos Chavez wrote: I see that I can purchase G.729 licenses for my Asterisk server, but I have seen that many phones support a G.729 variant like A or B. Are these suppoted by the same G.729 codec in Asterisk? -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] -Stable CVS-03/19/04-04:37:11 Not working properly
I just compile -Stable cvs but when I dial an extension I can't hear the phone ringing, but the phone is really ringing and if someone picks it up you can talk with the person but i cant really listen when the phone is ringing sip to sip didn't try anything else Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Important: The Asterisk Mailing list (new subject)
a news group could be less flood Miguel On Fri, 2004-03-19 at 19:00, Andrew Kohlsmith wrote: How about using a web form for posts instead of replying to an e-mail? A How about not. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura click click bad quality
no it wont happend with zap cards or other sipphones such as grandstream and wisip. Miguel On Tue, 2004-03-16 at 09:10, Senad Jordanovic wrote: Miguel Cavazos wrote: hello guys heres my setup i have 2 asterisk servers in 2 different houses sipura ulaw --- asterisk --- iax2 (ilbc) --- asterisk -- sipura ulaw, this is my setup but when i call the other sipura i can listen like click click click click it doesnt seem a bandwidth issue because it has dedicated dsl lines on both ends im using on both sipuras the lastest firmware .31 This could be DSL line noise? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura click click bad quality
if it was related to the dsl line i would notice my other phones such as grandstream and the ones on zap cards with the same problem im only having this issue with sipura. Miguel On Tue, 2004-03-16 at 12:00, Senad Jordanovic wrote: Miguel Cavazos wrote: no it wont happend with zap cards or other sipphones such as grandstream and wisip. I am referring to noise DSL service produces on the line. It is a very tiny but it it is there... So.. May be somehow it transfers into your IP network... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura click click bad quality
is this fixed on cvs -stable branch?? Miguel On Tue, 2004-03-16 at 23:02, Andres wrote: Miguel Cavazos wrote: hello guys heres my setup i have 2 asterisk servers in 2 different houses sipura ulaw --- asterisk --- iax2 (ilbc) --- asterisk -- sipura ulaw, this is my setup but when i call the other sipura i can listen like click click click click it doesnt seem a bandwidth issue because it has dedicated dsl lines on both ends im using on both sipuras the lastest firmware .31 my sip.conf looks like this [101] type=friend secret=mysecrete context=master-6186 callerid=Mike 101 host=dynamic reinvite=no canreinvite=no dtmfmode=inband transfer=yes nat=0 disallow=all allow=ulaw and iax.conf looks like this [oficina] type=friend username=oficina secret=mysecrete auth=plaintext context=asterisk host=dynamic disallow=all allow=ilbc this setup is for one of the servers the second server looks very alike to this could you help me out??? ill appreciate a response even i know your not consern about asterisk but i just get this problem with sipuras. Sounds like the Timestamp issue. Restart asterisk with DEBUG messages on and see if you catch these kind of messages when the Sipura line is experiencing bad quality: rtp.c:950 ast_rtp_raw_write: Difference is 720, ms is 110 If so then look at BUG: http://bugs.digium.com/bug_view_page.php?bug_id=0001195 Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPC5000 (WIP-5000 from hitachi cable)
thanx for the review michael, could you send some pictures of the phone? can you tell how long does the battery lives? signaling what do the menus have how do you configure it etc? maybe after you do a full testing we can do a Wisip vs. IPC5000 working futures. Miguel Cavazos On Mon, 2004-03-15 at 10:25, Michael Devenijn wrote: My review for this wireless SIP phone Got it friday tested it this weekend with asterisk ... everything went well (could just place one call) and then i've got authentication faults but i think it's a asterisk isue that could be solved ... i've changed the authentication settings on asterisk to accept anything and everything went well (simply placing a call, reciving a call) But saturday (after charging the batteries), nothing ! I just could start it up but then the software seems to be blocked completely and i couldn't find any software reset or upgrade ... Some major issues : - couldn't see on the screen when a call was missed - couldn't transfer a call (except with the asterisk #) - phone loses registration, need to push the reg key after 10 min) (here stops my test with the software side of the phone due to the lock) - some buttons (8 and down) don't work properly ... - it is a early test unit, this phone is not really for the market see software version (0.0.2) which says enough about the buggy state of the software. but if they are working on the software (a lot) it would be a great phone very light weight, very clear screen , good useability Kind regards Michael Devenijn -Oorspronkelijk bericht- Van: Craig Waddington [mailto:[EMAIL PROTECTED] Verzonden: ma 15/03/2004 11:07 Aan: Michael Devenijn CC: Onderwerp: FW: IPC5000 FYI _ From: FahdTel AB [mailto:[EMAIL PROTECTED] Sent: 12 March 2004 17:47 To: Craig Waddington Subject: RE: IPC5000 Hello Craig, Thank you for your kind inquiry. Pls find attached handset and pricing information. Best Regards Mohammed -Original Message- From: Craig Waddington [mailto:[EMAIL PROTECTED] Sent: den 11 mars 2004 19:04 To: [EMAIL PROTECTED] Subject: IPC5000 Hi, I am looking to purchase some of these phones. Can you provide me with information and prices please. Thank you, Craig Waddington. DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura click click bad quality
hello guys heres my setup i have 2 asterisk servers in 2 different houses sipura ulaw --- asterisk --- iax2 (ilbc) --- asterisk -- sipura ulaw, this is my setup but when i call the other sipura i can listen like click click click click it doesnt seem a bandwidth issue because it has dedicated dsl lines on both ends im using on both sipuras the lastest firmware .31 my sip.conf looks like this [101] type=friend secret=mysecrete context=master-6186 callerid=Mike 101 host=dynamic reinvite=no canreinvite=no dtmfmode=inband transfer=yes nat=0 disallow=all allow=ulaw and iax.conf looks like this [oficina] type=friend username=oficina secret=mysecrete auth=plaintext context=asterisk host=dynamic disallow=all allow=ilbc this setup is for one of the servers the second server looks very alike to this could you help me out??? ill appreciate a response even i know your not consern about asterisk but i just get this problem with sipuras. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPC5000 - Wireless Sip phone
you buy the unit thats what its call a test unit ipc5000 looks great and its 28 USD more than wisip i think the lcd is worth Miguel On Thu, 2004-03-11 at 19:58, Craig Waddington wrote: Thanks for the info. Sounds good. Does that mean I can contact them for a test unit also, to try before I buy? __ From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn Sent: 11 March 2004 18:25 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] IPC5000 - Wireless Sip phone I ordered a test unit and will recieve it this week (already shipped from sweden), i will post some comments on this list when it is tested .. I hope it will do his job !! ... the mail they sent to : Hello Michael, Hope you are well. Your sample is on the way and pls find attached delivery note for your reference. Ps. frieght charge was USD10 lower, so we own you USD10 that we will pretty reduced it with your next order or we transfer it to your bank account. I'll the coming days send you updated information about the handset and its new design i.e. it has L2 roaming feature now. The handoff time is 200 ~ 300ms between the AP. We aim to short it to 100 ~ 200ms. The implementation of Web Authentication (web-login) what we call HTTPS(SSL) is ongoing and should be released on June. It can be software upgrade. Best Regards, Mohammed Fahd -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Craig Waddington Verzonden: do 11/03/2004 19:15 Aan: [EMAIL PROTECTED] CC: Onderwerp: [Asterisk-Users] IPC5000 - Wireless Sip phone I am looking to buy a wireless sip phone, probably the IPC5000, I have looked at Wisip phone and read tons of posts regarding that phone. Do any * admins have any feedback on this phone? Is there any major differences between the phones, besides looks? The site has very limited information regarding prices etc. Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPC5000 - Wireless Sip phone
The problem with the few wifi sip phones is the size, its to big compared to normal cellphones. I got a wisip and i was going to buy an IPC5000 but he has no paypal or credit card charge online store and so But the point is wisip and IPC5000 are about the same size both the big difference is that wisip has a small lcd and IPC5000 has a bigger lcd, this could make a difference in a future for webbrowser on the phone or msgs to irc,msn,icq what ever. A webbrowser could help roaming alot since many wifi providers like starbucks use web auth as a login. I think wifi phone are the future and both wisip and IPC5000 are the first generation of wifi phones also the expensive cisco wifi phone that doesnt support sip yet. So i expect in a near future smaller phones with a high battery life. In my expirience with a wifi sip phone you can go about EVERY where and use it school, office, house, friends even on the street if you have a palm with netchaser to search wifi hotspots or a small wifi detector thats about 20 dlls or a lap on the car you can find many Access points (about 2 every 150meters) where you can stop and make a call ( a call wont harm the owner of the access point ). I had several call this week from access points on the street and I think that the voip future is wifi. However the battery on wifi phones wont be so good for now since many wifi devices eat batterys in a couple of hours. Buy IPC5000 the big lcd is worth the extra 28USD Miguel Cavazos On Thu, 2004-03-11 at 18:15, Craig Waddington wrote: I am looking to buy a wireless sip phone, probably the IPC5000, I have looked at Wisip phone and read tons of posts regarding that phone. Do any * admins have any feedback on this phone? Is there any major differences between the phones, besides looks? The site has very limited information regarding prices etc. Ta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BCM Wireless SIP Phone
the phone works for the wlan600 its a great phone poor battery but even palms with wifi use ALOT more battery when wifi is on and considering this phone has the wifi ON all the time the 23 stand by hours and 3 hr talk is ok it registers with asterisk just fine, try get it from pulver Miguel On Wed, 2004-03-10 at 04:51, Steven Thomas wrote: Hi, Has anyone tried this Wireless SIP phone with Asterisk? If so, any limitations? Thanks. http://www.bcm.com.tw/product/productIS.htm Regards, Steven Thomas Network Integration Services IBM Australia Ph: 0404 099 262 NH011, IBM Centre, 601 Pacific Hwy, St Leonards, 2065. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Palm OS5 client
ive been looking for a palm os5 client found gphone there webpage claims to be sip but i just cant make it register against asterisk Miguel On Thu, 2004-03-04 at 07:05, Dean Collins wrote: Does anyone know of a Palm OS5 client that can connect to asterisk? Hopefully I can use gprs to connect back to my home pabx and make local calls while on the road. Also can anyone comment on how well the CE clients work? Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wisip firmware, updates, features??
hi guys finally i got my wisip this week and im very happy with it. It works but i was wondering anyone know where can i find new firmware, updates or a wish list? I cross emails with jeff pulver about having a small http browser for auth on starbucks hotspots mcdonalds or prodigy movil(mexico). Even to check some text things via web maybe email??? He seems not to be so intrested so ill try emailing the manufacture. However if someone has a useful url or can tell me where to find this information please send me an email. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi Phones
yes it does and g723.1 also, im about to buy it, i will be sending feedback to the list as soon as i get my unit. I really like alot how it looks, hopefully i will also love how it works:) Miguel On Mon, 2004-02-16 at 12:57, HQ wrote: Miguel, IPC5000 doesn't support G729 (8 kbps) (it only support G711 64kbps) Be carefull with what you buy. Hector. - Original Message - From: Miguel Cavazos [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Sunday, February 15, 2004 9:38 AM Subject: [Asterisk-Users] Wifi Phones Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wifi Phones
Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details you can find it here http://mike.calle69.net/IPC5000-brochure.PDF, and its almost the same price as Wisip. But when I ask if this phone will work with asterisk I got this answer We didn't tested on Asteriskt but on IX66, hotsip, Cisoc etc... However, The IPC5000 should work on other SIP platform without any problem as it is standard based. I just dont want to spend 290 USD for a phone that wont work and that no one seems to use here. So I would like to know if anyone of you guys had try out this model or seen it working, sorry about the unnesesary traffic to the list, my question is simple would this work against asterisk if anyone knows any other Wifi phones besides Wisip and Ciscos expensive toy please tell me. Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voiceglo questions, IAX
how many simultanius calls does voiceglo permit??? Miguel On Fri, 2004-02-06 at 01:28, Cameron Palmer wrote: IAX is what they use with glophone. http://webphone.voiceglo.com. It is a seperate server from the myphone.voiceglo.com SIP gateway. The IAX gateway is msps01-nyc.voiceglo.com on port 5036. cameron. On Thu, 5 Feb 2004, Jim Flagg wrote: - Original Message - From: Michael Swan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 05, 2004 5:23 PM Subject: [Asterisk-Users] Voiceglo questions 1. Can someone confirm whether Voiceglo needs to use SIP or can it handle IAX? This link seems to indicate it uses SIP: http://www.mail-archive.com/[EMAIL PROTECTED]/msg20561.html although other messages on the mailing list indicate that Voiceglo is using Asterisk in its internal architecture. Brian West indicated in this post http://lists.digium.com/pipermail/asterisk-users/2003-December/029076.html that he had Asterisk registering using IAX. Can Brian or anyone else post a copy of their IAX.conf Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2
would be a good idea to put it on the changelog, i see its there but it doesnt really inform nothing. Miguel Cavazos On Wed, 2004-02-04 at 15:21, Mark Spencer wrote: Asterisk 0.7.2 is now released and contains lots and lots of bug fixes from the bug tracker. Highly recommended for people running 0.7.1. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller
same here, when i recive an incoming call from x100p to line 1 on sipura, i can hear them but people can't hear me im using 1.0.24 on my firmware Miguel On Sun, 2004-01-25 at 20:54, Chris Higgins wrote: Frankie Gravato wrote: I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse Service and DID's when i get Phone call using the Voicepulse or Pstn the caller can't hear me or barely hear me. The Sipura is running Firmware 1.20 and calls are being passed using Ulaw Codec? Anyone out there in the asterisk community please oh please help me before i do something that my asterisk server won't like. I just received my Sipura on Friday and have been testing it extensively over the weekend. I have noticed an issue similar to what you mention above. For the record, the sipura tells me I'm running software version 1.0.20. Also, there is NO nat configuration that is causing my problem. When I receive a call over my X100P and dial my 3 SIP phones (one gs budgetone 100, two analong phones through sipura), if I answer the analong phone connected to line 1 of the sipura, the caller cannot hear anything. I've only noticed this problem in this exact scenario. The other situations listed below have no problems whatsoever and audio works in both directions: 1. Call from sipura line 1 to any internal SIP phone. 1. Call from any internal SIP phone to sipura line 1. 2. Call from sipura line 1 out through X100P. 3. Call into my X100P from outside and answer sipura line 2. 4. Call into my X100P from outside and answer sipura line 2 and THEN transfer to sipura line 1. 5. Call into my X100P from outside and answer sipura line 1 (the caller cannot hear audio for this leg of the conversation), TRANSFER to any other line, and transfer back to sipura line 1. After the second transfer, the caller can hear audio from sipura line 1. I don't know what is special about line 1. I've switched my analog phones across the two ports on the sipura to make sure it wasn't one of my phones (not that I thought it was anyway). Frankie, have you tried the same experiment, but pulled your analog phone from line 1 and put it in line 2? Has anyone else seen issues like this with line 1 on a sipura? Thanks.. -- Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth ? + Doc + cdr
check http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html this email has help me alot with BW and codecs Miguel On Mon, 2004-01-12 at 08:13, WipeOut wrote: Hans-Henrik Andresen wrote: Hi, How much bandwidth do I need for 1 conversation ? I know it depends on the codecs, in X-lite I can see a codec called gsm, and the grandstream aha analog/ip converter have a codec called 721. G.711 will use ~84Kbps and GSM will use ~34Kbps.. Doc. I have found the asterisk handbook, but only a draft from marts 2003 anything newer ? Yes the handbook is a little old now but will still give you a good foundation.. Guides/howtos are welcome as well. http://www.digium.com/index.php?menu=documentation Look at the unofficial links section on the bottom for some good info.. anyone have a php interface to accounting ? Nope not that I know of.. you would have to develop your own.. :) Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SpeakFree
sad, yes but who needs speakfreely when you have asterisk and soft/hard phones. The author seems really unmotivated so let him find the path maybe he could join asterisk development team :) Miguel On Sun, 2004-01-11 at 19:33, Lists wrote: If you did not see slashdot today, check out this anoucment. http://www.fourmilab.ch/speakfree/ Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive
The following errors occurred during your registration: * The username you entered as your referrer could not be located. cant create a username Miguel On Sun, 2004-01-11 at 22:53, Brancaleoni Matteo wrote: Hi http://www.asterisk.bz Alternative to the asterisk-users list nothing against this forum, but this made me think. I noticed that some people loose their time in setting up doc sites... the idea is great, but since there're already grown sites (oej's wiki), why not stopping into doing something already done and spend that time writing docs, for example ? matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] analog or sip ? was far end disconnect supervision
sip phones have alot of nice features and they really work, you can try some phones under $200 yes, but about the analog phones, people like to have there cordless phones, or there micky mouse phone or garfield phone so consider that. You loss some features but your customers get the phones they want to have in there room, office, kitchen, living room, etc. Besides you can get cool atas under $100 USD GS or sipura. Sip phones get old and look ugly, analog can be replace at any moment. Miguel On Sun, 2004-01-11 at 23:07, Lance Arbuckle wrote: Thanks to everyone that responded to my channel bank question. Ive decided that the Adit 600 would be a good choice. Then I got to thinking about SIP phones and wondered if their quality has progressed to the point that they can be deployed to customers who just want their phones to work and wouldn't tolerate any SIP hickups. As for pricing, I would think the SIP phones would need to be in the $200 price range to be competative with analog or ADSI phones plus a channel bank. I know there are lots of variables that figure into the analog vs SIP question like number of incoming lines and how they're delivered and the number of extensions etc I guess what would be helpfull to me would be some general rules of thumb that you asterisk experts use to determine what type of extension phones to recommend for a given customer. Thanks -Lance ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial from command line?
you need a sound card on the box that works Miguel On Thu, 2004-01-08 at 20:54, SamW wrote: I have 2 installations of asterisk. On CLI one server has Dial command. Other installation do not have Dial command on the CLI. What I am missing. How to enable dial command from the CLI. Thanks, - SamW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Inexpensive Analog Ports
2 sipuras SPA2000, sold at 100USD each they have 2 FXS ports its like cisco ata Miguel On Mon, 2004-01-05 at 07:38, Asterisk Newbie wrote: Does anyone know of any inexpensive alternatives to the four port analog module offered by Digium ($305) what work seamlessly with asterisk? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Scaleable Solution for small office
on the 5 to 8 secs of echo if your using on your zapata.conf echotraining=yes you should get rid of this echo problem Miguel On Tue, 2004-01-06 at 19:15, Ryan R. Fligg wrote: Hi, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo from the X100P cards is completely irritating to my users. I have tweaked the system as much as I can, IE: switching tip and ring, using aggressive suppressor, tweaking the TX/RX settings. There is still echo in the first 0-8 seconds of the call while the echo cancellation is catching up. I understand the problem is that you do not hear this echo in analog systems even though it is present and it is only heard in the digital system because it is not as fast as analog. So here is my problem. Our CEO wants me to get rid of the system unless I can provide a solution that will expand and work just as good as an analog replacement PBX. He came to me with this decision after trying to use the Snom200 speaker phone for a conference call, not a good idea as we all know the speaker on this phone is not a conference phone replacement. He also was frustrated when he tried to patch a call that came to another extension, which transferred fine, but was disconnected when the receiving party made a switch on their PBX to another extension and it was dropped. Currently we have 4 POTS lines and I have 1 setup for DSL and the other 3 hooked into the X100P cards, 2 outgoing and 2 incoming, using the DSL as incoming rollover. We are planning on expanding our business pretty rapidly and he wants a system that will be easy to setup and scale. I know asterisk and VOIP phones are great for this but the little glitches in the phones, hardware wise, are not supporting my Asterisk decision. I would like to upgrade to a channel bank, and was wondering if anyone has had any echo issues with other digium hardware. I know that the X100P issue comes from not having ECAN DSP in the card. I was also wondering if anyone had any luck adding a conference phone such as the Polycom Soundstation Conference Phone. Any suggestions would be appreciated Ryan R. Fligg Secured Digital Storage, Inc. 104 SW 4th St. Des Moines, IA 50309 Phone: (515)-244-6290 Cell: (720)-841-5802 Website: www.dstorage.com E-Mail: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk reload for FWD to register
hi, guys I have a weird, really weird problem with FWD (free world dialup), My serveris a P2 400mHz 64 on Ram. This server is setup only to answer to its FWD # for a friend to do calls to its local PBX. When i boot up the server running debian Woody, with kernel 2.4.23, running asterisk CVS from DEC 30 2003, asterisk is loaded with no problems on the init.d, so asterisk runs fine but it doesnt show attempts or try nothing similar to register with FWD, i have to manually log in the console and reload asterisk by reload, after that it shows its trys to register with FWD and finally makes it and recives incoming calls from FWD. The problem here is I don't want to reload the server every time the box goes for reboot so I wonder if someone is having this same problem or had it before. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Quality Survey.... :P
On Wed, 2003-12-24 at 08:18, Brian Capouch wrote: I have about a dozen Budgetone 101s and I'm pretty much satisfied with them. Sorry, Brian; they've got some rough edges, but they're $65, for God's sake. They are $65 yes, but you can get the best analog phones on the market for that price and use an ata. If GS could give the information for people on asterisk to develop iax this $65 phone could be even better than most of the phones in the market more features less buggy and cheaper than all the other sip phones out there Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merry Christmas, all Asterisk users!
merry xmas olle and you all in the list happy holidays!!! Miguel On Tue, 2003-12-23 at 23:04, Olle E. Johansson wrote: It's the day before Christmas here in Sweden, actually the night before at this time... We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into merry-christmas-mode with the yet undocumented CLI command frosty-mode on, a mode where the PBX will connect all incoming extensions to the ho-ho-ho sound file and then randomly pick a number in the +1234 country code (for the North Pole), dial out and bridge. And these magical SIP connections will work over ANY type of NAT. (Due to the SIP header Santa-magic-cookie: on) And yes, the frosty mode is even un-documented on the http://www.voip-info.org wiki. :-) It's been fun spending the fall with the Asterisk project. I look forward to next year, with the new handbook coming in place, with many new applications and features and - hopefully - many new Asterisk installs at customer sites. It's snowing outside, the trees are already covered with snow and the stars are glittering on a dark sky. My kids are sleeping, dreaming of their christmas gifts tomorrow. It's going to be a traditional Swedish christmas... Have a wonderful Christmas, all of you! Warm regards, /Olle ___ 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] Grandstream Quality Survey.... :P
brian i hate to do this but you miss something. The little ata286, well people hear ECHO when they talk to me they hear small echo with the ata in use the caller id works 75% of the times maybe a little less with the lastest firmware calls get drop with no reason after some time, hopefully they can give out information for this IAX firmware Miguel Cavazos On Wed, 2003-12-24 at 05:43, Brian West wrote: Today class we are going to be talking about the wonderful line of GrandStream products. Or should I say BarbieTone phones? Who else is having MAJOR issues with the grandstream products? How many times have you been told upgrade upgrade upgrade? How many of you have paperweights, granted the phone is light as a feather and couldn't weight papers down in the first place? How about that ring tone, really dandy eh? Who else is irked about the the GAPS crap? It should slurp down cfgMACADDRESS.txt and we shouldn't have to pay more for that option. Have you had Message Waiting Indicator issues? Have you had issues with the Hold button and flash button? Have you had issues with sip transfers? Has the grandstream product line made you want to hurt someone? Care for some matches and lighter fluid? Was the response from grandstream support able to take care of your problems? I own a grandstream phone and I guess I just don't use it enought to see alot of these problems but the consensus on #asterisk is they are CRAP and everyone should stop buying them till they get their act together. A few people in the asterisk community have offered to write IAX firmware for the phones but grandstream has give them the run around. If they can't create stable and usable firmware they should atleast let the info out to let someone write IAX firmware for the damn thing. BOYCOTT GRANDSTREAM Thanks, bkw_ PS: then again you get what you pay for, 10 dollar phone with a 65 dollar pricetag. ___ 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] Grandstream Early Dial
On Thu, 2003-12-18 at 04:03, Brian West wrote: Stop using beta firmware... I honestly think that GrandStream needs to either fix the phones or stop making them.. THEY SUCKS! I think I would rather eat glass than work with a grandstream phone. bkw Brian, GS has people that works very hard on this BETA firmwares, and if you have any problem with there phones send them the tcpdump logs, asterisk logs, and explain the bugs your having. THEY are really really open mind to listen up their firmware problems. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Headless Linux system for Asterisk
well i never had a asterisk server with a monitor or keyword all my servers i do remote login with ssh its better more private. Miguel On Thu, 2003-12-18 at 22:02, Michael Welter wrote: Because of space limitations and because of the location of the punch-down blocks, my * server is located on the shelf in a coat closet. Sadly, there is not enough space (or ventilation) for the monitor and keyboard. This will all change when we move to new quarters, but... Does anyone have experience running Linux/Asterisk without a monitor? What, if any, are the issues? TIA ___ 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] IP 500/600 1.1.0 Firmware
good to hear theres going to be support for this phones, but why not put it on the wiki??? so we can have all the faq in one place. Miguel On Mon, 2003-12-15 at 22:23, mattf wrote: I just got off of the phone with Scott Willard at Polycom and things seem promising. He's going to send me the latest stable boot firmware for the Soundpoint 500/600 phones and I will make that available on a webiste somewhere for people to download. It seems like their big stumbling block to giving us the firmware and admin docs for their phones is that they don't want to have to support end users of their phones. They usually have their resellers do support for specific PBXs that their phones work with and because none of their resellers support Asterisk or sell Asterisk that means no one supports Asterisk users of their phones. I told him that we would be putting up a support website for their phones that they can point Asterisk end users to, to download the latest firmware and the admin docs as well as contact people for help with configuration. He seemed to accept that idea and is going to send me a copy of the latest bootloader as a sign of goodwill for stalling for so long. I am to present him with a document outlining what kind of support we will provide and how they can point end users that call polycom to us so that we can help. I plan on putting up a simple website with a few FAQs and sample configs as well as some people that can be emailed or called to ask questions about Polycom on Asterisk. I'd love to have a few volunteers for that list so I don't get swamped :) I plan on hosting the website(donated by my company of course) at least in the near term just to get something up. Let me know if you're interested in helping out. I'll post the new bootloader on a web site when I get it. Thanks, MATT--- -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Monday, December 15, 2003 4:01 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] IP 500/600 1.1.0 Firmware Hello, I've talked with various people at Polycom about getting on their developer program and they have been stalling me for weeks. The most information I received was from Scott Willard, head of IP phone sales for Polycom. He said that the company is not sure of what direction they want to head with selling phones to end users in the IP-phone area(and letting their firmware and admin manuals out in the open). They are planning to continue development of the IP phone line and not get into the PBX area. I sold the virtues of Asterisk to the best of my abilities and he seemed to be interested in at least the possibility of working with the community, but he said the final decision would be made by the IP-phones division head of development. When he did not say. Scott also shed a little light on Polycom relationship with other IP phone companies, he said that Polycom licenses a lot of their speakerphone and other phone technologies to many companies including Cisco and Avaya but their hardware is made entirely differently than those companies, the resemblance to the Cisco phone line is purely coincidental. MATT--- -Original Message- From: Bisker, Scott (7805) [mailto:[EMAIL PROTECTED] Sent: Monday, December 15, 2003 3:10 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IP 500/600 1.1.0 Firmware Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb ___ 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] WiSip phone experiences?
thats an old review jeff pulver said firmware was better now, is it worth 250USD? im thinking of one does it autodetect ssid? works with low signal? sound quality? Miguel On Sun, 2003-12-14 at 18:34, John Todd wrote: http://www.loligo.com/asterisk/misc/WiSIP/ Works decently enough. Still some software bugs to work out. JT At 10:10 AM -0600 12/12/03, Michael Graves wrote: Anyone here have any good/bad things to say about first hand experience with the new Wifi SIP phones? I am considering one for my office as an alternative to FXS+Analog cordless. Thanks, Michael Graves -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] FWD 54245 I've been everywhere man, I've been everywhere. - Hank Snow ___ 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] GS early dial
it doesnt work here, same firmware 4.26 tryed it with 4.18 also and it doesnt work, i press any number and it gets screw up i will try it with the handytone ata286 and see if it works, anyway its the same firmware but its worth to try out Miguel On Sun, 2003-12-14 at 21:13, John Breeden wrote: Am I assuming that a GS set to early dial to * dosn't work. Or am I missing something? Tried inband, info and rfc288, all nojoy. I'm assuming that it's not/supported or GS bug, only asking because it's assumptions that alwas get me :-) GS firmware 1.0.4.26 Thanx in advance John Breeden Hawaii ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
it's a firmware problem on GS, they are working on that but it seems its not that simple to make volume higher on the speaker and echo go away, anyway 4.26 seems stable for now and with many new features! Miguel On Fri, 2003-12-12 at 17:55, Bob Knight wrote: John, did you ever get any feedback from the GS wish list? I love the BT-102's with 1 exception. The speaker phone. I have not come up with a combination that makes it acceptable. If I had a way to cover up that button I would go ahead and deploy the phone. But the db level and echo to the far end user makes it unusable. If anyone on the list has successfully configured and used the GS speaker phone, could use please share thanks, bk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
biggest is DSL can now be connected directly to the GS no need of dhcp or static ip no more, mute/del button works now the date tap on the screen and many more cant remember download it at http://www.grandstream.com/TEMP/FIRMWARE/ or update with 4.3.153.50 tftp Miguel On Fri, 2003-12-12 at 19:47, rnc Info Lists wrote: it's a firmware problem on GS, they are working on that but it seems its not that simple to make volume higher on the speaker and echo go away, anyway 4.26 seems stable for now and with many new features! Miguel, What are the new Features? Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk freezing HELP
im using 1.0.4.24 and i havent seen any problems yet! http://www.grandstream.com/TEMP/FIRMWARE/ Miguel On Tue, 2003-12-09 at 04:11, TeleSIP wrote: We are a service provider and are constantly being handed new firmware by Grandstream to test on our network. Every time we ask when it will be offically on their web site they answer soon. But a week or two later there is a new version that needs to be tested. (and so far all versions have failed one or more of our tests) It seems that the latest firmware has a bug that was not there on previous releases. It is not very reassuring for us to still see bugs that we have reported over 3 months ago. Andres. - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 08, 2003 10:29 PM Subject: Re: [Asterisk-Users] Asterisk freezing HELP TeleSIP wrote: And by the way, do not use firmware 1.0.4.18 on GS phones. It contains a nasty SIP Port bug. Where do you guys come up with this Grandstream firmware that isn't on their website? Are their back channels that the rest of us are kept out of? This is the umpteenth reference to firmware in the 1.0.4 series, and all their TFTP server is handing out is 1.0.3.81. What's going on with that? Thx. B. ___ 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] x100p/hangup detection issues? - SOLVED!
i have this very same problem but i have a different context
; Answering incoming calls
[incoming]
include = asterisk
exten = s,1,Wait,15
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Background(${SOUNDS}/casa)
exten = 1,1,Dial(SIP/101)
exten = 2,1,Dial(SIP/152)
here my problem is they can dial any extension so at what time should i
put the ringing stuff?
Miguel
On Fri, 2003-12-05 at 06:45, Patrick Cantwell wrote:
OK everybody, I have solved my problem!
The issue lies in how you handle the incoming call. I actually stumbled
across this while trying to find a better way of doing fax autodetection.
The trick is you *must* use Answer to pick up the call. The *WRONG* way of
doing things is:
exten = s/,1,Zapateller
exten = s,1,NoOp
exten = s,2,PrivacyManager
exten = s,3,Goto(100,1); proceed to home phone
rules
exten = s,103,Hangup() ; In case caller doesn't
supply info correctly, hangup
(of course omit the zapateller/privacymanager stuff if you're not using it)
The *CORRECT* way of doing things is:
exten = s/,1,Zapateller ; if CID is not present,
send telemarketer blast down the line
exten = s,1,NoOp ; if it is, don't do
anything :)
exten = s,2,PrivacyManager ; check for CID. If not
present, prompt caller for their number.
exten = s,3,Answer ; pick up the call. this
allows asterisk/x100p to correctly detect phone company signalling!
exten = s,4,Ringing; generate some ringing
for the calling party
exten = s,5,Goto(100,1); proceed to home phone
rules
exten = s,103,Hangup() ; In case caller doesn't
supply info correctly, hangup
Note the rules 3 and 4, Answer and Ringing -- they do what you'd expect.
Answer takes the call from the PSTN and Ringing makes asterisk generate a
ringing tone while it handles the call internally. This allows it to a) let
callers know the call is still in progress, and b) allows the fax rule to
pick up a fax machine during ringing! I couldn't find this documented
anywhere, or I would have implemented it this way from the get-go. Hope
this helps someone else! (Let me know if it does!) Maybe this should be on
the wiki too? :)
Thanks,
Pat
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 1:22 AM
Subject: Re: [Asterisk-Users] x100p/hangup detection issues?
On Thu, 4 Dec 2003, Jonathan Tew wrote:
We're testing with an X100P card. When the caller on the POTS line
hangs up it never causes our IAX phones (DIAX in this case) to hang up.
Curious what you find out.
a) try kewlstart if you're lucky enough to have it
b) try BUSYDETECT
c) alternatively in the US try callprogress
- wasim
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Re: [Asterisk-Users] XBOX as and * Dedicated Server
On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote: During Phreaknic, Mark was showing off a Xbox running asterisk with 4 S100U interfaces connected to the game ports on the front. It was interesting. In the end, I don't think it is cost effective as a real PC since you can also build a PC of similar or better specs for that price now and you get PCI slots. the S100U is a good idea, and yes you can get a pc for what 30bucks a P200, but i was looking for something small and good looking, i dont have a big room and another CPU. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XBOX as and * Dedicated Server
On Fri, 2003-12-05 at 11:17, Senad Jordanovic wrote: I like you idea. Very Cool :) Is RAM upgradable on xbox? Thanks no its not, BUT its very optimize the xbox hardware should work REALLY REALLY GOOD i dont know how good it will be on long uptimes Miguel Cavazos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XBOX as and * Dedicated Server
about the hardware the xbox has HD 8 GB, ethernet port and 4 usbs a nvidia video card and all you need is the usb adapters and some hours reading on how to break in without a mod chip using an xbox its a cool idea because its nice and small and cheap! using the S100U was a great idea for the pots Miguel On Fri, 2003-12-05 at 16:23, Joseph Finley wrote: Can you imagine trying to sell this to a customer though? The customer see's you walk in the door w/ an X-Box or PS2 and they say That's our phone system ?!? Not that I would think someone would do it, just a thought that entered my mind. Joe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Friday, December 05, 2003 11:15 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] XBOX as and * Dedicated Server At 07:17 AM 12/5/2003, you wrote: I guess for the XBox you would need some external gateway. Audicodes or Mediatrix come to mind but they start at $500. A year ago, I installed Linux on Playstation 2. I had to purchase it with the hardware for about $200. (40GB, keyboard and some network adaptor). It actually worked. And its a much more open community than Xbox. Now that you mention it I will revisit this. http://playstation2-linux.com/ I have asterisk running on a $300 Soekris motherboard. Works perfectly. --Ernest ___ 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] XBOX as and * Dedicated Server
Hello guys, i have been on this mailing list for some weeks now, and i was wondering if someone here has installed linux on the XBOX and use it as a dedicated server. Its a 200 USD computer and could make it perfect to asterisk, its little and doesnt really take much space. My question is could this make it for a stable server??? here are some links i found for linux on XBOX http://xbox-linux.sourceforge.net/ some intresting screenshots found on that URL http://xbox-linux.sourceforge.net/docs/screenshots.html The only real thing that i dont know is where am i going to put the X100p. Miguel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does Asterisk use CPU?
if you have 80calls going, its time to think on getting a good dedicated server, switches, for the work and UPS with big batterys also some good power supplie:) Miguel On Fri, 2003-11-28 at 12:51, Matthew Asham wrote: Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Thanks Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FYI: Simple Small Asterisk install..
try base instalation of debian delete the documentation and asterisk sources should be less than 150megas all Miguel On Wed, 2003-11-19 at 16:12, John Todd wrote: At 3:41 PM + 11/19/03, WipeOut wrote: Hi, If anyone is looking for a small Asterisk installation I have managed to get it down to 296MB (If you remove the kernel source code.. could probably be made smaller if some of the devel packages and asterisk source is removed as well.) To do it I used Trustix Secure Linux 2.0 (http://www.trustix.net), did a minimum install with ssh support(92MB) and then added the required packages individually.. Everything has compiled and Asterisk loads but I haven't tried any Zaptel drivers or hardware so can't comment on that.. Anyway if anyone needs a small secure install there it is.. Later.. I would imagine that if you could get it into 256mb, that would be ideal, since it would fit into many flash disk chips, while 292mb means one would need to buy a 512mb chip (much more expensive.) JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wifi600 or other Wifi sip phones
http://www.pulverinnovations.com/ On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote: Where can I buy the Wifi600 phones ?? Or does anyone know of other Wifi SIP phones ?? Any help would be appreciated Regards Mick ___ 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] Wifi600 or other Wifi sip phones
anyone knows if this phones now support auth with sid ??? my school wireless lan needs the auth Miguel On Tue, 2003-11-18 at 21:57, [EMAIL PROTECTED] wrote: Where can I buy the Wifi600 phones ?? Or does anyone know of other Wifi SIP phones ?? Any help would be appreciated Regards Mick ___ 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] Unable to find path from G729A to ULAW on Sipphone.com
i followed what you said didint work heres what console says i cant do the 1800 call anyway -- Executing Macro(SIP/101-8376, callerid-pstn) in new stack -- Executing SetVar(SIP/101-8376, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-8376, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-2e46 is making progress passing it to SIP/101-8376 -- SIP/fwd-2e46 answered SIP/101-8376 == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-8376' -- Executing Macro(SIP/101-c43c, callerid-pstn) in new stack -- Executing SetVar(SIP/101-c43c, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-c43c, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-bc38 is making progress passing it to SIP/101-c43c -- SIP/fwd-bc38 answered SIP/101-c43c == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-c43c' On Tue, 2003-11-18 at 21:58, Barton Hodges wrote: [EMAIL PROTECTED] wrote: I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A Before somebody tells me UTFG, I ALREADY HAVE. Somebody else had a similar issue last week and there was no real resolution posted. So here it is again. I have all of the codecs that I support enabled in my sip.conf. Here is the relevant section: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw ; Allow codecs in order of preference allow=gsm allow=ilbc register = 17476692375:[EMAIL PROTECTED]/1101 [sipphone] type=peer username=17476692375 secret=[MYSECRET] host=proxy01.sipphone.com fromuser=SteveSokol fromdomain=sipphone.com canreinvite=no ; ==END OF SIP.CONF FILE=== The issue occurs whenever any calls that route over the sipphone peer are made to a toll-free number. The calling phone (either my GS100 or my X-LITE softphone) rings two or three times then gives me busy. Here is the entire debug output: -- Executing Dial(SIP/1101-1f83, SIP/[EMAIL PROTECTED]|20|tr) in new stack -- Called [EMAIL PROTECTED] NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1234379840]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A -- SIP/sipphone.com-e7b3 is making progress passing it to SIP/1101-1f83 -- SIP/sipphone.com-e7b3 answered SIP/1101-1f83 -- Attempting native bridge of SIP/1101-1f83 and SIP/sipphone.com-e7b3 NOTICE[1242768320]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1242768320]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/4) == Spawn extension (default, 918884510851, 1) exited non-zero on 'SIP/1101-1f83' The problem does NOT occur when I call another sipphone.com user (i.e. GS100 - Asterisk - Sipphone - GS100). Those calls go through just fine. The toll free calls were working last week. Is it me, or is it Sipphone.com? Any suggestions would be greatly appreciated. Steve I've been having the same types of problems (I'm probably the guy you're referring to who had the same problems last week). This is the solution I have found to work reliably so far. Configure the Grandstream BT101 with the following codecs, in the following order: choice 1: G.729A/B (g729) choice 2: PCMU (ulaw) choice 3: PCMA (alaw) choice 4: G.729A/B (g729) choice 5: PCMU (ulaw) choice 6: PCMA (alaw) Configure the codecs in
RE: [Asterisk-Users] Unable to find path from G729A to ULAW onSipphone.com
-- Executing SetVar(SIP/101-a9e5, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-a9e5, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] == No one is available to answer at this time On Tue, 2003-11-18 at 22:50, Barton Hodges wrote: [EMAIL PROTECTED] wrote: i followed what you said didint work heres what console says i cant do the 1800 call anyway -- Executing Macro(SIP/101-8376, callerid-pstn) in new stack -- Executing SetVar(SIP/101-8376, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-8376, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-2e46 is making progress passing it to SIP/101-8376 -- SIP/fwd-2e46 answered SIP/101-8376 == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-8376' -- Executing Macro(SIP/101-c43c, callerid-pstn) in new stack -- Executing SetVar(SIP/101-c43c, SIP_CODEC=g729) in new stack -- Executing Dial(SIP/101-c43c, SIP/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] -- SIP/fwd-bc38 is making progress passing it to SIP/101-c43c -- SIP/fwd-bc38 answered SIP/101-c43c == Spawn extension (asterisk, 18006927753, 3) exited non-zero on 'SIP/101-c43c' You need to modify the lines in extensions.conf to match your configuration: Try this: exten = _1800NXX,1,SetVar(SIP_CODEC=g729) exten = _1800NXX,2,Dial(SIP/[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
