similar to: Cisco FXO as PSTN gateway (updated request for assistance)

Displaying 20 results from an estimated 4000 matches similar to: "Cisco FXO as PSTN gateway (updated request for assistance)"

2004 Jan 12
1
Cisco FXO as PSTN gateway
I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see
2004 Jun 23
1
Asterisk user/host registration
Hi Folks, I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server. When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below. *CLI> sip show peers Name/username Host
2004 Jul 25
2
Incoming SIP gateway context?
I just started service with Broadvoice.com and everything seems to work. However, apparently my understanding of incoming sip contexts is less then what I thought it was. Could someone point me in the right direction? (* on a public address, CVS-HEAD-07/12/04, C7960 phones) In my sip.conf I have: [general] port = 5060 bindaddr = 0.0.0.0 allow=ulaw tos=0x18 ;sets ip tos bits (=lowdelay and
2004 Jan 23
1
PSTN incoming - both SIP & H323 always arrive in default context :-?
Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 & get exactly the same result (both for chan_h323 & chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for
2018 Feb 26
4
How to update modules in iniramfs fastly
> -----Original Messages----- > From: "Steven Tardy" <sjt5atra at gmail.com> > Sent Time: 2018-02-26 10:48:48 (Monday) > To: "CentOS mailing list" <centos at centos.org> > Cc: > Subject: Re: [CentOS] How to update modules in iniramfs fastly > > On Sun, Feb 25, 2018 at 8:29 PM wuzhouhui <wuzhouhui14 at mails.ucas.ac.cn> > wrote:
2005 Jul 05
4
Asterisk on Linksys WRT54G
Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk
2003 Aug 10
3
Asterisk Newbie ...
Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long
2003 Nov 15
2
Internal server error - cannot align media streams - help needed
Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP
2004 Aug 09
1
Inbound Call Errors...
I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30
2004 Apr 02
7
Welltech FXO: initial tests
Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to
2003 Sep 07
2
New cvs compile; basic operational question, please.
Can someone offer a hint on what I'm doing wrong with the basic * config? Just implemented * for the first time using yesterday's cvs. The initial configs are based on John Todd's article at http://www.onlamp.com/lpt/a/3956, and using two 7960's for initial testing. When one 7960 calls the other, I get the following and the call is dropped: -- Executing
2004 Aug 21
0
autocreatepeer and sip peer options
Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there
2004 Sep 08
0
re: asterisk, SER and autocreatepeer
Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of
2003 Nov 18
2
ask problem about softphone--asterisk--softphone, Urgent!!!
Hi, all, I want to use asterisk SIP as a proxy, and two softphone (Ubiquity SIP Phone) as user agent, like below: Softphone1<-------------->Asterisk SIP<------------>Softphone2 (User Agent) (Proxy) (User Agent) 155.69.xx.xx 155.69.yy.yy 155.69.zz.zz zhou mysipproxy.com
2005 Jul 01
1
no voice
Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and
2004 Nov 12
1
Shorewall''s bogon file needs updating
As far as I can tell from <http://shorewall.net/errata.htm> the current shorewall bogons file is <http://shorewall.net/pub/shorewall/errata/2.0.8/bogons> which contains the line: 58.0.0.0/7 logdrop # Reserved This is incorrect. These two /8s were allocated to APNIC as of April 2004. See also <http://marc.theaimsgroup.com/?l=nanog&m=108319003517919&w=2> and the main
2005 Jan 03
2
PSTN to VoIP FXO gateways?
Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%
2006 Oct 25
3
Quintum DX as gateway to PSTN for Asterisk
Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When
2003 Dec 22
4
MSN to GS - Call drops in 10 secs
Hi All, i dont what changes i made recently but i am unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS
2010 Jul 25
2
R equivalent of SAS proc freq
Dear R-users, I am looking for a R function that would be the equivalent of the SAS proc freq ( http://support.sas.com/documentation/cdl/en/procstat/63104/HTML/default/viewer.htm#/documentation/cdl/en/procstat/63104/HTML/default/procstat_freq_sect006.htm). The table, ftable, xtabs functions are close but do not quite offer the same capabilities (e.g. they just return counts and no %ages as far as