Re: [asterisk-users] Zaptel vs DAHDI
I am using the asterisk 1.4.22 and dahdi 2.0.0 I dont have the module oslec installed I am using the default echo cancelation: mg2 whene I used the option aggressive=1 the echo been worse any idea thx in advanced From: Tzafrir Cohen tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Sent: Thursday, December 25, 2008 4:05:16 PM Subject: Re: [asterisk-users] Zaptel vs DAHDI On Thu, Dec 25, 2008 at 05:30:31AM -0800, Jacob Blonisci wrote: hello shalom what ia the alternative of AGGRESSIVE_SUPPRESSORin zconfig.h in the DAHDI version dahdi_echocan_oslec.ko ;-) Sreiously, though, it has been made a run-time option. In chan_dahdi.conf: echocancel = yes,aggressive=1 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.2 and openser 1.4
Hi: Can asterisk 1.2 and openser 1.4 work togather ? Regards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 and openser 1.4
At 07:32 AM 12/27/2008, fateme fatah wrote: Hi: Can asterisk 1.2 and openser 1.4 work togather ? Regards. Yes. I have done large deployments that where multiple SER (and OpenSIPS) are used for either inbound or outbound (supplier) proxy. with multiple ASTERISKS and Cisco 5400s. I have also put both on same linux box (5060 for Asterisk , 506X for SER) when necessary to meet technical challenges on interface with specific carriers. ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme - play the name
sasikala kala wrote: Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their name, but what i want is, if it gets the name one time, thats all, it should just play that name whenever the call comes from the same callerid. That's not a realistic expectation. How can you presume that because callerid is xyz that it's always the same person calling? You can not. Office's routinely have one main number with callerid being the same for all office users. I would not be surprised to find two users calling in separately from the same office having the same callerid, where you can not tell them apart based on callerid. Now having said that. I can see where you could/would be able to get the name announced once on arrival and get meetme to save that and announce they left and then forget the recorded name. If they were disconnected, they may need to re-record their name. I don't know if that is a feature now or not, but that would be doable and you could ask for a new feature based on this description. Lyle Is it possible to achieve this feature by some way? Hope somebody would have the same requirement, kindly help to achieve to do the same. thanks and regards Sasikala. Add more friends to your messenger and enjoy! Invite them now. http://in.rd.yahoo.com/tagline_messenger_6/*http://messenger.yahoo.com/invite/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.2 and openser 1.4
asterisk 1.2 , is enough old to make a lot problems, upgrade to 1.4 or 1.6 and enjoy it. integration opensips( ser) and astersik, is the best solution for the big voip systems. --- On Sat, 12/27/08, Mike Trest m...@trest.com wrote: From: Mike Trest m...@trest.com Subject: Re: [asterisk-users] asterisk 1.2 and openser 1.4 To: faza_...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: asterisk-users@lists.digium.com Date: Saturday, December 27, 2008, 5:04 PM At 07:32 AM 12/27/2008, fateme fatah wrote: Hi: Can asterisk 1.2 and openser 1.4 work togather ? Regards. Yes. I have done large deployments that where multiple SER (and OpenSIPS) are used for either inbound or outbound (supplier) proxy. with multiple ASTERISKS and Cisco 5400s. I have also put both on same linux box (5060 for Asterisk , 506X for SER) when necessary to meet technical challenges on interface with specific carriers. ..mike.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with DAHDI hangup on calling out.
I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri
1.4.7
and I am getting the error:
-- Requested transfer capability: 0x00 - SPEECH
-- Called 23/317506
-- Channel 0/23, span 1 got hangup, cause 99
-- Hungup 'DAHDI/23-1'
on DIALING out. Calling in seems to work just fine.
Seems like everything is configured fine...
system.conf
span=1,0,0,esf,b8zs
bchan=18-23
dchan=24
chan_dahdi.conf
[channels]
signalling=pri_cpe
switchtype=national
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
context=smvoice-incoming
group=1
channel = 18-23
extensions.conf in use is :
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
lspci shows
TE205P dual span card 5.0v
What might be going on here? Everything was working with zaptel 1.4.5.1
and asterisk 1.4.18 so my T1 is good.
Jerry
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Re: [asterisk-users] Audiocodes MP-11X configuration to work withAsterisk
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk also, I want all the channels to register with asterisk. I have the FXS channels working fine, I cant acheive that with the FXO channels, does anyone have any advice or possibly sample configs. Thanks in advance :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate and fax detection
Hi everybody, I have an application that uses Originate AMI command to initiate outbound calls. However, I cannot find any way of redirecting calls that were originated to a fax machine to some other extension (e.g. fax extension). Is this possible? Or is there any way to get info from the AMI that the originated call went to a fax machine? BR, Dex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate and fax detection
Perhaps AMD() supports this. Not necessarily reliably. On Dec 27, 2008, at 6:10 PM, Asterisk aster...@abraxas.si wrote: Hi everybody, I have an application that uses Originate AMI command to initiate outbound calls. However, I cannot find any way of redirecting calls that were originated to a fax machine to some other extension (e.g. fax extension). Is this possible? Or is there any way to get info from the AMI that the originated call went to a fax machine? BR, Dex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
