similar to: How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to *

Hi All, I have two SIP softphones(Windows Messenger) running on different subnet (Phone-1 on IP XXX.XXX.25.ABC & Phone-2 on IP XXX.XXX.15.XYZ) and my Asterisk Server is running on IP XXX.XXX.25.PQR.Because of some security issues both the subnets are completely isolated ( U cant even PING from one to other) and I want to connect Phone-1 & Phone-2 to the *. How can I proceed? Please

Hello, I am using chan_capi and would like * to behave differently depending on the MSN the caller dials. Is there a way to route incoming ISDN calls to different contexts based on the MSN dailled? I have tried something like msn=1234 incomingmsn=1234 context=msn1 msn=4567 incomingmsn=4567 context=msn2 in capi.conf but with no results. Thanks for any hints. -Walter -- Walter Doerr

Hi All, Many thanks to Marc who helped me with a previous Capi Dialout plan - however..... What I now would like to be able to do is: - We have 8 msn's 383590, 383591 383592 etc. What I would like to do is set up an If Then Else type statement along the following lines: - If extension 7957 Then Dialout on Capi msn 383590 ElseIf extension 7958 Then Dialout on Capi msn 383591 ElseIf

Hi all, I've finally taken the plunge and had isdn installed. It's a BRI from Telewest in the UK. I'm told by the provider that it's in point to multipoint mode with 6 digit presentation. I'm using an AVM fritz (bt speedway) with the fcpci-suse9.1-3.11-02 driver. Chan_capi seems to be installed fine. I can receive incoming calls. However, I can't call out. When ever i

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hi, We are using asterisk version 1.0.5. We have registered two UA's with asterisk. (Registration was successful) UA1 <-------> * <--------> UA2 Now, UA1 subscribes for UA2 to asterisk. asterisk sends NOTIFY to UA1 with UA2's state as open. But if UA2 gets un-registered then, asterisk is not sending NOTIFY to UA1. But when there is state change from UA2, asterisk is

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for

Yeah, I'd like to hear you guys' opinion instead of CleverNickName's! -----Original Message----- From: Luki [mailto:lugosoft@gmail.com] Sent: Tuesday, February 15, 2005 9:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Comments on Vonage SIP port blocking complai nts?? You can always visit Slashdot for countless (useless, well,

The simplest solution and the one already implemented in linux, tmpfs. It would be best to allocate 4-8GB to tmpfs on /tmp and let the kernel do the work it was designed to do. And you would not be limited to PCI bus speeds. The DDR2800 is about 12GB/sec. Some would say "overheads, etc, etc". Agreed, even at 95% loss (doubtful) you still get higher badwitch then PCI bus/hard rive could

As a general rule of thumb it would be good to make the distinction between 'Credit Card' and 'Debit Card' too. If possible, never ever use a debit card for online purchases. It taps directly into your account and removes REAL money. Credit cards are 'virtual' money in that they are credit and your cash in account remains untouched. That way if your credit card account

Hi everyone, I'm wondering I would accomplish the following: I want to dial several SIP extensions simultaneously, HOWEVER, for different times (say ext 10 for 15 sec and ext 11 for 30 sec), and potentially with different headers (such as ALERT_INFO) and codecs for each extension. Obviously whoever picks up first gets the call. After the longest timeout expires (30 sec in this example) I want

Hi all, does anyone have any luck using a Cisco 7971 (SIP) behind NAT with two different accounts on the same server (i.e. two different extensions)? I am using Cisco-CP7971G-GE/8.3.0 and asterisk V1.4.something. The phone sends SIP packets from a high-numbered UDP port but expects a reply on port 5060. Fine, I do some magic with iptables to rewrite the packets (which limits me to one phone at

On Feb 2, 2008 10:16 AM, Luki <looky at poczta.fm> wrote: > Hello Carlos > > could you please help me to solve issue with Mustek 1000VA UPS under > Freebsd ? > I am not able to connect with it : > > >Network UPS Tools - UPS driver controller > >Network UPS Tools - Megatec protocol driver 1.5.4 [megatec] > >Carlos Rodrigues (c) 2003-2007 > >

Outward dialing is a no brainer. VoipJet is the best outbound call provider I have come across. Period. It always works for me and the call quality is always very very good. So far that seems to be the norm for them. I am still working on getting my inward DIDs solidified so no opinion there... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Thanks. I know that line quality is a factor, and I know I could get a 50$ fax with a PSTN line (that is what I have now). But I have my reasons to want to setup a fax over IP, and I want to keep going. Where do I find info on this debug mode? Is there a detaild log in Asterisk that show exactly what happens when the fax is trying to come in? Also, could this console output help? - Executing

Hi all, I'm having this weird issue that I can't explain. Maybe someone can explain what is happening. This is a Asterisk install that has been in production for 6+ months. It's version 1.2.10. Couple weeks ago one SIP peer started disappearing randomly. And I mean it simply disappears. One second "sip show peers" shows it, and then it's gone. A simple "sip

Hi I try to compile samba-3.0-alpha19 on my Linux-Box (SuSE 8.0, Kernel 2.4.18, cc 2.95.3, make 3.79.1). After "make" I got this errors: Linking bin/smbd /usr/src/krb5-1.2.5/src/lib/libgssapi_krb5.a(accept_sec_context.o): In function `rd_and_store_for_creds': accept_sec_context.o(.text+0x7b): undefined reference to `krb5_rd_cred' accept_sec_context.o(.text+0xdb): undefined

I am running RH 7.3 with kernel 2.4.18-27.7.x. Recently I have added a new 60GB Quantum IDE hard drive to the server. I have set it up as a MASTER on a separate IDE channel. Two partitions on the /dev/hdc are /dev/hdc1, /dev/hdc2 - both using ext3. For more detailed HDD parameters, this is 'hdparm -i /dev/hdc': ----------------- /dev/hdc: Model=IC35L060AVV207-0, FwRev=V22OA66A,

Hi Darren/Damien, Sorry for responding so late. Still hope we can get this sorted out. Yes I am indeed using PAM for ssh authentication and disabling priv seperation is a no-go for us since it opens up a security loophole. From what I can see in ptree and auth logs, when the child passwd process returns with SIGCHLD, the parent sshd process terminates. Sshd logs are as follows as requested at