Re: [asterisk-users] 3Com 3102 Phones

?? Thanks On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp

Re: [asterisk-users] 3Com 3102 Phones

On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp

[asterisk-users] 3Com 3102 Phones

Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I

[asterisk-users] redirect based on incoming number

How does one redirect calls based on incoming number or caller ID or the lack thereof? current I have for number 123-4567 that it redirects all 800 , 877 and 866 numbers to Voicemail directly. If the primary area code is 352 then accept this and pass it to extension exten =

[asterisk-users] rxgain / txgain for iaxmodem or hylafax

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be heard I can read about rxgain and txgain for zapata. my fax setup

Re: [asterisk-users] Asterisk and Faxing

Barry Fawthrop wrote: Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need

[asterisk-users] Asterisk and Faxing

Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes

Re: [asterisk-users] Asterisk and Faxing

Kevin P. Fleming wrote: Barry Fawthrop wrote: I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I

[asterisk-users] Good 4 Port PSTN Gateway

Hi All I have a simple network with 2 IP phones, 1 Std. Phone (using a Siptura 2000) and a CG-410 Gateway to connect the the two PSTN lines that I have. I have a odd hassle that for no apparent reason, the calls will quite working. but quite I mean the phones will ring but their is no voice

Re: [asterisk-users] Dialplan checkup

line through the TDM card to the fax machine? Right, this is possible also or use an ATA if not a TDM card ?/ Thanks All Barry Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard

Re: [asterisk-users] Dialplan checkup

Hi Gordon Following you dial plan How does Asterisk know to move from s,2, to either incoming,1, or fax,1, The only jump I recognize it Goto(internal,incoming,1) which should take all calls to incoming,1, and not fax,1, OT: is spandsp rxfax handled by astlinux ? Thanks again Barry Gordon

[asterisk-users] Dialplan checkup

Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten =

Re: [asterisk-users] Dialplan checkup

,.. and what make it go to exten = fax, How does this logic work?? Thanks again Barry Barry Fawthrop wrote: Hi All Curious will this work Std. PSTN line ---x-- X100p | -- Fax Machine Using a standard home phone pstn line

[asterisk-users] VPN As SIP Tunneling?

Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting VoIP Traffic Say a VoIP User and VoIP Termination Provider. Each side can put QoS onto their part, but if QoS does NOT exist between them then call

Re: [asterisk-users] Recommendations for QoS, PoE Switches

Hi David Care to share how you approached using Diffserv and VLANs with the FSM7326P We are considering the same switch. But I'm unsure about the configurations required. Thanks in advance Barry David Coulson wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Typically we deploy the

Re: [asterisk-users] VPN As SIP Tunneling?

will be relevant to OpenVPN, which might be a bit different from IPsec, PPTP or other solutions. Am Montag, den 11.12.2006, 17:26 -0500 schrieb Barry Fawthrop: Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting

[asterisk-users] Use of VPNs

Hi all Is the use of a VPN between IP-PBX and VoIP Provider a useful tool? Since the QoS and general traffic of the Internet can never be predicted, would the implementation of a VPN between Client and VoIP Provider increase voice quality and/or security or is the converse true ? Thanks

[asterisk-users] VLANs and Quality

Hi all How much does configuring a network with VLANs improve or effect quality ? Is there much reason to justify the configuration of VLANs ( I know networking, but not VLANs at all) Would it not be better to find high traffic users and determine why? Your Thoughts Thanks Barry

Re: [asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

what is the sip.conf for 1239 which I'm going to assume is a extension on the TNT Barry JR Richardson wrote: Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip sip TNT pri pri asterisk The TNT is running 11.0.6 and the asterisk

Re: [asterisk-users] Tampa Bay Asterisk Users Meetup on Monday

Hi Kristian How many are you going to have at tonight's meeting ? Thanks Barry Kristian Kielhofner wrote: Matt Florell wrote: Hello, We will be having another Tampa Bay Area Asterisk Users Meetup on Monday, November 6th at 7:30 PM. Asterisk users from gurus to new users are welcome. Along

[asterisk-users] Caller ID 1.2.10

Hi All Has there been problems with Caller ID and Asterisk 1.2.10 ??? I have a Phone Number from Teliax, I get this chan_sip.c:10468 handle_request_invite: Failed to authenticate user 3523029577 sip:[EMAIL PROTECTED];tag=as03efd979 And the call does not go through. I have a CG-410 FXO

Re: [asterisk-users] AstLinux 0.4.4 Released!

Hi Kristian What else is in the 0.4.4 release ?? Any news on the Sangoma A200 or Faxing ? Thanks Barry Kristian Kielhofner wrote: Hello everyone, I have released AstLinux 0.4.4. Thanks to all of the testers on astlinux-users, AstLinux 0.4.4 now includes mISDN support (again). Grab

Re: [asterisk-users] AstFax Sending a Fax

On 10/24/06, *Barry Fawthrop* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi All I'm trying to understand how I would send my fax ? If I use Word or what ever word processor or even an email client to create what I want faxed. I have *asterisk setup with and FXO

[asterisk-users] AstFax Sending a Fax

Hi All I'm trying to understand how I would send my fax ? If I use Word or what ever word processor or even an email client to create what I want faxed. I have *asterisk setup with and FXO Gateway that will make the call to the fax number I dial SIP extension 320 is the FXO gateway.

[asterisk-users] Clipcomm CG 410 / FXO Gateway

Hi all Anyone got a Clipcomm CG410 working with Asterisk ??? Where incoming calls are passed on directly to asterisk and outbound calls are passed directly to PSTN ?? I have tried configuring the gateway and I just can't get it to register no matter what I try If any could share

[asterisk-users] Call Quality / Echo / Problems

Hi all I'm having a problem getting usable quality from my Asterisk setup. *SETUP* 2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones in the house and 2 x FXO to receive calls and in the future faxes. Gentoo Linux Here is what I've done so far (1) Moved theTDM 400p (FXS,

[asterisk-users] Asterisk behind Sonicwall firewall

Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port ? Why ? Thanks all ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Asterisk behind Sonicwall firewall

this nothing was changed it blink I'm not going to work now Thanks all Barry Dr. Michael J. Chudobiak wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been and then suddenly it drops UDP packets because SIP is no longer on port 5060 but some random assigned port

Re: [asterisk-users] Asterisk behind Sonicwall firewall

all the VoIP SIP setting are enabled/configured and SIP is packet forwarded to the Asterisk Box due to Sonicwall NAT Now I'm trying to find out why and how to correct this. Thanks all Barry Rich Adamson wrote: Barry Fawthrop wrote: Hi all Anyone using a sonicwall firewall ? I have been

Re: [asterisk-users] Asterisk behind Sonicwall firewall

Hi Francesco Yes it is SonicWALL Enhanced has an option called 'Persistent NAT'... Is it turned on? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Cisco 7960 Double Natted

Thanks for the input Yes I have nat=yes and qualify=yes I know in the SIPMacAddress.cnf file I have # NAT/Firewall Traversal nat_enable: 1 nat_received_processing: 1 nat_address: phone's public IP Address Do I still need to set it again in SIP Configuration ? Thanks all Barry Hughes, Sam

[asterisk-users] Cisco 7960 Double Natted

Hi All Yes I know double Nat is a problem But I have a Cisco 7960 which is remote from the * PBX ad connected via the Internet. Each side has NAT (1) Sometimes it will work often it won't. And when it decides to work is random Always (2) The Register side works fine. SIP SHOW PEERS has

Re: [asterisk-users] Cisco 7960 password reset

If the phone already had the SIP image running. Check the SIPDefault.cnf file there may be a phone_password= string this is the phone's current password use it remember to change to number or uppercase if need be Ferguson, Michael wrote: Maxx, Thanks much for the feedback. I will check

Re: [asterisk-users] SIP Connection Problems

behind NAT. You can try setting port forwarding on the phones side as well as look at a better router. Some routers will make you pull your hair out while others will work almost perfectly (this explains my now bald head :) ) Dovid - Original Message - From: Barry Fawthrop [EMAIL

[asterisk-users] SIP Connection Problems

Hi All I have a Cisco 7960 which is connected remotely to an Asterisk server. Both are unfortunately behind NAT. The Phone registers and is show in sip show peers, with the correct public ip for the phone and a 100ms qualify time (1) I can dial the phone from another phone, it will ring but

[asterisk-users] Using a DB for Configurations

Thanks David But what I was more looking for was storing the configuation file eg extensions.conf as a database file in MY SQL and then have asterisk load the table from MYSQL as opposed the text file extensions.conf ? Is there any benefit in this ? Thanks all Barry

[asterisk-users] Using a DB for Configurations

Hi All Is there a benefit to using a database to hold the extensions and sip .conf information/configurations or is using the standard Text file just as good and no benefit is received? Also how does one go about converting the text .conf files to a database, and the have asterisk read it

Re: [asterisk-users] New Asterisk GUI

Tried to install and get this extension_dir does not exists /usr/lib/php/extensions/no-debug-non-zts-20020429 when entering index.php Any Ideas ? Barry El Flynn wrote: Hello, We've just released our Libero Management System application, a web-based interface to configure and manage your

Re: [asterisk-users] Provider UNREACHABLE

Thanks All for your replies A couple have mentioned backup routes, as I'm clueless on backup routes How do I program the dialplan in extensions.conf to: (a) try multiple provider to make an outgoing call based on current latency between my * box and the different providers ? (b) have if

Re: [asterisk-users] Provider UNREACHABLE

Thanks All First off I never mentioned Teliax (but yes correctly ASSUMED they are my provider) and this is not a Teliax issue per se My issue is more the fact that I have Qualify = yes in sip.conf but repeatedly get REACHABLE and UNREACHABLE as can be seen below. even when I set Qualify =

[asterisk-users] Provider UNREACHABLE

Hi All I am repeatedly getting a UNREACHABLE and then REACHABLE about 10 sec apart most of the time and then sometimes for about 45 - 74 minutes I have tried a reload and sip reload but neither bring the provider back ? What else could I try and how do I prevent this Thanks in advance

[asterisk-users] Rate or rank ITSP

Hi There I know of wiki there is a list of VOIP providers, but is there a list or can we create / suggest one that will list VoIP providers, their location and quality of service ? Too me this will be very valuable, plus looking at some of the requests of late I'm sure others would like that

Re: [Asterisk-Users] echo / delay problem

I'm in the US, using cards bought direct from Digium. I have lowered the rxgain and txgain to -8 and that seems to be helping futher. I wish I could understand why? The problem with more time is that I can hear myself in the headset of the std. phone as well as the party on the other end. The

[Asterisk-Users] echo / delay problem

I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was

[Asterisk-Users] Learning the Ropes of *

In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module 2) Is it best to use a TDM

Re: [Asterisk-Users] Re: Snom and multiple lines

I'm also trying multiple lines on my phones I have mapped P2 - P5 numbers 2502 - 2505 I have 2502 - 2505 in my sip.conf andextensions.conf snom200 is using 2.05e 1] the LCD for p2 - p5 are always lit. When I dial 2503 the MWI flashes none of the p1 - p5 flash :(I was

[Asterisk-Users] * will not load, after latest CVS install

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] = (Skinny Client Control Protocol (Skinny))May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541

Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk

canreinvite=no [general] port = 5060 srvlookup = yes nat = yes tos = lowdelay disallow = all allow = ulaw allow = gsm allow = alaw context= INVALID Currently my IP phones haves this in the sip.conf [4403] type= friend username

Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk

The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf:

Re: [Asterisk-Users] IAX and Snom200

- Original Message - From: Christian Stredicke To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 10:05 AM Subject: RE: [Asterisk-Users] IAX and Snom200 We thought about this option. I guess the IAX2 is not the problem. We believe the real

Re: [Asterisk-Users] IAX and Snom200

NAT traversal is a huge issue I agree with Michael and Brian what with the latest viruses etc... security is and will be more and more of an important issue, many SOHO and small corps. Often don't have the know how or finanical backing to implement standard/conventional security and internet

[Asterisk-Users] Failure SIP / RTP

Greetings I'm now really messed up::: setuptrying to by pass NAT *1 (bound to LAN addr)- Linux Gateway + *2 (bound to public addr)- Internet - IP Phone A *1 and *2 are registered and see each other fine *1 has ext 13 SayUnixTime *1 has ext 4403 an IP Phone on the LAN From IP Phone A I

[Asterisk-Users] IAX and Snom200

Greetings What would it take to have a snom200 support IAX, what are the processes or having hardware to support a new codec? Can this be tested and done by a uesr or must this be done by the manufacturer? Thanks in advance

Re: [Asterisk-Users] Snom 200

Progress It seems I can't hear the Say Time, due to RTP Double NAT I'm guess this is both problem 1 and 2 really issue. My config: IP Phone - Router (Nat) - Internet - Linux (NAT) - * Server ANyone know of work arounds the double NAT? or other methods to route RTP with snom 200s, to work

Re: [Asterisk-Users] Snom 200

Thanks to All who replied I have tried all the steps above. and from the website given I have two snom 200 next to each other 4403 and 4405 when I dial 4405 - 4403 nothing rings and * CLI reports voicemail/default/4403/busy when I dial 4403 - 4405 nothing rings and * CLI reports

Re: [Asterisk-Users] Snom 200

Thanks Willy and others It seems I am not able to make myself clear on my two problems I wish to try again, as I'm sure it is not the phones, but some stupid config problem on my part. I repeat alot of what I have said in order to try make myself clear. I have a 4401, 4403 and 4405 in sip.conf

[Asterisk-Users] Snom 200

Greetings All I'm busy trying out my new snom 200(s) I have it connected and * CLI tells me registered 1) I pick up the handset and hear the dial tone 2) Dial and Ext, that says Date Time (13) 3) * CLI scrolls that the call is connected and time is being spoken YET the handset is quite and

Re: [Asterisk-Users] Store caller IP in CDR

From: Olle E. Johansson [EMAIL PROTECTED] snip Check the CDRuserfield - it's a free field in the CDR you set in the dialplan or from a script. How would you set the CDRuserfield from the dialplan exten = ? Thanks in advance B ___

Re: [Asterisk-Users] Snom 200

From: [EMAIL PROTECTED] Please include the sip.conf entry for the phone you have .. SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 192.168.0.15 externip = 24.73.215.62 localnet = 192.168.0.0 localmask = 255.255.255.0 tos = lowdelay disallow = all allow =

Re: [Asterisk-Users] Snom 200

Here's another funny * CLI puts put -- Registered SIP '4405' at IP.address Port 5060 Expires 3600 and within seconds the snomm 200 beeps the MWI goes on the LCD and the light flashes a call from asterisk Not Found Willy if you could let me see you sip and config files, if you have yours working?

Re: [Asterisk-Users] NuFone?

I would like to know how and where nufone and the other get their access to provide termination, and yet only offer 2.9c /min We are about to sign up with a termination provider and have read often that people suggest nufone, yet for everyone who suggest them, they always have had some sting in

Re: [Asterisk-Users] Maximum retries exceeded on call

check NAT setting try taking it out of sip.conf, that worked for me Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

Re: [Asterisk-Users] Maximum retries exceeded on call

check context perhaps try include in the extensions.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] * and PrePaid

Greetings What would it take (all hardware etc..) to setup * on a prepaid card server. I have an * server a T1 and TDM10B card, thus allowing 24 simultaneous calls I guessing I need a VoIP Termination Provider (eg: NuFone, etc..) How do I print and create the cards, and what are the

Re: [Asterisk-Users] Big Install examples please

Even though it was 100, I'm also keen to hear about large installs, what kind of experience did you have setting it up, and what hardware for the * server did you use? Thanks in advance Barry Matthew wrote: I've set up 75 extensions... I'm 100. Sorry. Sincerely,Matthew

Re: Spam Alert: [Asterisk-Users] Is IAXtel down?

Hi Matt How do you reach a Iaxtel 1-700 from PSTN? didn't think this was possible ? Barry

[Asterisk-Users] Asterisk and Faxing

Greetings Can a dial plan be set so that an * server connected to a T1 can receive a fax and save it as a file in a specific dir or have the fax sent to a specific IP address, that will handle the fax and save it as a file? Thinking along the lines of fax - email, but within an * server,

[Asterisk-Users] Asterisk Faxing

What is the best or simplest method to connect 4 fax machines into a * system? Fax - ATA-186 - Switch - * Server - VoIP or PSTN Fax - * Fax Server with TDM 400P - Switch - * Server - VoIP or PSTN Would like a dedicated # on the T1 to do direct to the fax machine. Would love your

Re: [Asterisk-Users] Asterisk Faxing

- Original Message - From: James Golovich [EMAIL PROTECTED] James Golovich wrote: I don't explicitly disable echocancellation on the channels I use for fax, and zaptel always seems to detect the tone to disable echo cancellation from the fax. I send/receive all my faxes over IAX2

[Asterisk-Users] Firefly and IXATEL

Hi How do I set up firefly to use the IAXTEL network? I have registered with IAXTEL I have a username and password. I have tried IAX protocol and iaxtel.org ad the network and nothing happens ? Thanks barry