[asterisk-users] Need help connecting Alcatel 4400 PBX to Asterisk

Title: Need help connecting Alcatel 4400 PBX to Asterisk Hi there I have a TE110P card fitted in my linux box running : Linux version 2.6.9-5.ELsmp ([EMAIL PROTECTED]) (gcc version 3.4.3 20041212 (Red Hat 3.4.3-9.EL4)) #1 SMP Wed Jan 5 19:30:39 EST 2005 I followed the installation steps

Re: [asterisk-users] FXO Cards vs. Channel bank with T1

Dovid B wrote: Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO cards ? Thanks. channel bank is more friendly to faxes and modems (v90 can work too) ___ --Bandwidth and Colocation provided by Easynews.com --

RE: [asterisk-users] Example Polycom function key config

Hi Jamie - Hi Noah, Has anyone here reprogrammed their Polycom features keys using sip/ipmid.cfg? If so I would be really grateful if someone could send me an example Here's the keys line that I use for one of my clients: keys key.scrolling.timeout=1

[asterisk-users] SIP realtime issues

Does anyone see anything wrong here? CLI realtime load sipusers name 1000 Column Name Column Value id 1 name 1000 callerid "Don" 1000 host dynamic nat yes disallow all allow gsm type friend context inbound secret 41674

[asterisk-users] Manager API - Originate Call - Need Help

Hi all,How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number?I can originate a call from my SIP-network using this parameters in Originate call command :Channel = SIP/0041435215301Context = defaultExten = 00982166501553Priority =

Re: [asterisk-users] Manager API - Originate Call - Need Help

On Wed, 2006-11-01 at 03:23 -0800, Ehsan Khosrowshahi wrote: Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command :

Re: [asterisk-users] Manager API - Originate Call - Need Help

If I understood your question correctly, you just need to reverse everything. Channel = OUTGOING TRUNK i.e. ZAP/00982166501553 Context = default Exten = internal extension that points to - 0041435215301 Priority = 1 CallerID = 0041435215301 This will first initiate the call to the number

[asterisk-users] Help me on Call parking

Hello Users...I'm Strucked in Call parking...I'm Using the Asterisk-1.1.11 version in My FC5 box,In That there is feature.confI'm Using SIP channel By using Asterisk + OpenSER [general]parkext = 9006 ; What extension to dial to parkparkpos = 9007-9009 ; What extensions to park calls on. These

[asterisk-users] ${CALLERIDNUM}

Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external calls. exten = ,1,GotoIf($[${CALLERIDNUM}

[asterisk-users] Re: Manager API - Originate Call - Need Help

In article [EMAIL PROTECTED], Ehsan Khosrowshahi [EMAIL PROTECTED] wrote: How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command :

[asterisk-users] IAX Realtime MD5 authentication

Hi, Is there any possibility to have md5 encoded passwords in the IAX users database? I notice the secret AND/OR md5secret columns always have to contain the password in plain text even when you set the auth column value to md5?!? Am I missing out something? Any ideas on how to correct this?

Re: [asterisk-users] ${CALLERIDNUM}

The following will: exten = s,1,GotoIf($[${LEN(${CALLERID(num)})}=2]?50) On 11/1/06, Scott Pinhorne [EMAIL PROTECTED] wrote: Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if

Re: [asterisk-users] Help me on Call parking

raviprakash sunkara wrote: In Extension.conf .. I'm confused to give the Dial planning.. You don't need to do anything in the dial plan for parking. Just transfer the call to your parking extension and Asterisk will take it from there. Doug

Re: [asterisk-users] Re: Newbie Questions

Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to

Re: [asterisk-users] ${CALLERIDNUM}

On 11:53, Wed 01 Nov 06, Scott Pinhorne wrote: Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external

Re: [asterisk-users] ${CALLERIDNUM}

On Wed, 2006-11-01 at 11:53 +, Scott Pinhorne wrote: Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and

[asterisk-users] AEL2 - CUT function usage

Hi, In Asterisk 1.2.7, my AEL code looks like this: macro callForwardHunt(numargs,numlist,typelist,ttr) { for(x=1;${x}${numargs}+1;x=${x}+1) { CUT(number=numlist,-,${x}); CUT(type=typelist,-,${x}); NoOp(${number});

Re: [asterisk-users] siemens hipath interoperability - PRI/Q.SIG - cardrecommendation

Hi Wendy, I got this info from digium developers, that caller id name transfer/display (asterisk/iphone - pbx/clasic phone)) using ISDN/Q.SIG should work, so, do you have possibility to confirm this, if it realy working in practice (with siemens hipath idealy)? thanks PJ

Re: [asterisk-users] ${CALLERIDNUM}

Scott Pinhorne wrote: Hi Does anyone know how I can check if a callerID is more than 2 digits. I am setting up my phones so that if the callerID is 3 digits the phones ring one way if it is more than 3 digits it rings another i.e. internal calls and external calls. exten =

Re: [asterisk-users] ${CALLERIDNUM}

That worked great Many Thanks -Original message- From: C F [EMAIL PROTECTED] Date: Wed, 1 Nov 2006 06:57:28 -0600 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ${CALLERIDNUM} The following will:

[asterisk-users] wav format isn't compatible with Windows Media Player

Hi, When playing a wav-format (( low compression),(wav49-format?)) file with Windows Media Player, it plays the file and then sometimes bombs out with an error about how the file is corrupt or unsupported. If you listen to the file in wavepad you will hear the whole file, in Media Player the

[asterisk-users] Re: Asterisk and Panasonic KX Model

Thanks for help me with this issue. I've this scenario, a PANASONIC KX domain and an ASTERISK domain, each one with their own pool of extensions, incoming calls are recived by the PANASONIC KX as a gateway from PSTN to the office. Once a call is recived by the PANASONIC,it bridge the call to

[asterisk-users] a2billing

Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of

Re: [asterisk-users] Snom or Cisco Phones?

I love the Snom phones as well. The function keys are great and easy to use.On 10/31/06, mitcheloc [EMAIL PROTECTED] wrote:My vote is definitely for Snom, I've worked with Cisco phones foryears, but the Snom is much better integrated, and the feature buttons can be retooled for any environment,

[asterisk-users] a2billing

Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of

RE : [asterisk-users] Fxo box for asterisk ?

Hello, All the biggest gateways manufacturers do that. Search for Aliwei, Audiocodes, Patton, etc... Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Noc Phibee Envoyé : lundi 30 octobre 2006 20:51 À : Asterisk

Re: [asterisk-users] T.38 faxing with spandsp and Grandstream HT.486

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Johann Steinwendtner schrieb: I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? I've successfully

Re: [asterisk-users] Snom or Cisco Phones?

http://www.aztech.com/prod_iptelephony_ip150.htmlaztech rawks... the lcd has backlighting and methinks is snom inside ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Neat Application for Text to Speech

I was reading Octobers online edition of Wireless Asia and came across a company called CDyne (www.cdyne.com) They build a number of web services applications but among other things they have an application which you can fill out your details on a web page will some time in the future

RE: [asterisk-users] FXO Cards vs. Channel bank with T1

This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an FXO/FXS card. Ejay Hire -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Sent:

RE: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject

That looks like a rebranded Snom 300 to me. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rosli SukriSent: Wednesday, November 01, 2006 10:16 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [SPAM HEADER] - Re: [asterisk-users] Snom or

RE: [asterisk-users] Help me on Call parking

Hello. In extensions.conf; in the context that is dialed by your internal extensions, add this line. include=parkedcalls This will include the extensions created by the extensions module, and create your extensions 9006-9009. Good luck, Ejay Hire From:

Re: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject

it is, the navigation button is exactly the same, also notice the extreamly short handset cord On 11/1/06, Cory Andrews [EMAIL PROTECTED] wrote: That looks like a rebranded Snom 300 to me. Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]

RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no

Re: [asterisk-users] compilation problem with asterisk-addons

Erick Perez wrote: Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make

[asterisk-users] Which IP phones have best voice quality, preferably under $150

Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 andLinksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones,

[asterisk-users] Asterisk manager

I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the commands I am trying to send. Asterisk Call Manager/1.0 Action: login

RE: [SPAM HEADER] - RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones? - Email found in subject

Ken - take a look at using IAX protocol to route calls between your Asterisk boxes. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Williams Sent: Wednesday, November 01, 2006 10:58 AM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000

[asterisk-users] Upgrading from 1.0.9 to 1.2.6

Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines)

[asterisk-users] DTMF over IAX

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time.

RE: [SPAM HEADER] - [asterisk-users] Which IP phones have best voice quality, preferably under $150 - Email found in subject

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zeeshan ZakariaSent: Wednesday, November 01, 2006

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

The quality issues you describe may not be the fault of the individual phones! The quality of the PSTN connection, and the hardware through which it's connected can play as big a part in this scenario. The quality of the internal network with 50 IP Phones could also be part of the problem.

Re: [asterisk-users] ${CALLERIDNUM}

At 03:53 AM 11/1/2006, you wrote: exten = ,1,GotoIf($[${CALLERIDNUM} = ]?5) This will tell it to jump to 5 if callerID if but how do i tell it do jump based on length of callerID? exten = ,1,GotoIf($[1${CALLERIDNUM} = 1999]?5) Ira

Re: [asterisk-users] Snom or Cisco Phones?

Writing this as a user of VoIP and not a reseller, (meaning off the record), we really love the Snom phones here as well, I wish the Snom 300's had a bit more functionality (like the Grandstream), and the Snom 320's and 360's were a little less confusing with their buttons (aka too many buttons on

Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

I tend to stay away from the Grandstream phones for business use because they simply break to easily. I would suggest using Snom phones like the Snom 300 for around $99.2 Asterisk boxes in different locations? Sure, you can do that and its quite easily. On 11/1/06, Ken Williams [EMAIL PROTECTED]

RE: [asterisk-users] Asterisk manager

Sorry, but I failed to mention that I am running Asterisk BE B 1-1 I am trying to send commands to Asterisk manager via a telnet session. I am able to lo in and receive event logs from AMI, but when I try to issue commands I get an invalid/unknown command error. Here are some of the

Re: [asterisk-users] Audiocodes MP-114 noise

Jason,There are a couple things we can try to fix your problem.Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmpLet's try some quick things first though:In your web interface, go to advanced config - channel settings / voice settingsThere are some

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000

Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

Matt wrote: Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest version, but anything above .6, at this time, seems to have stability issues (I've tried them on multiple machines)

AW: [asterisk-users] Which IP phones have best voice quality, preferably under $150

snom 300 : CS -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kristian Kielhofner Gesendet: Mittwoch, 1. November 2006 12:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Which IP phones have best

RE: [asterisk-users] Which IP phones have best voice quality, preferably under $150

This e-mail, including attachments, contains privileged and confidential information intended only for the use of the addressee(s) name above. If you are not the intended recipient of this e-mail, or an authorized employee or agent responsible for delivering it to the intended recipient,

Re: [asterisk-users] DTMF over IAX

The problem is voicepulse, but they refuse to accept responsibility. From What phone are you pressing the DTMF?On 11/1/06, Jason Walker [EMAIL PROTECTED] wrote:Ok sorry for not being specific.I am having a problem when people outside call in to my number which terminates at VoicePluse then

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

I have the Budgetone 101 and GXP2000 and thought the sound quality was excellent. Even over the internet... I agree with Joe that something else may be the factor... Todd Zeeshan Zakaria wrote: Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000,

[asterisk-users] Re: DTMF over IAX

I had the same problem trying to use an iaxy for an overhead paging system. SIP has an option to set DTMF to inline, but iax does not. There was nothing I could do to get the iaxy to play audible DTMF tones. I had to use a SIP ATA for my paging system with the inline DTMF option. Note: The

Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

Ken, Also stay away from Swissvoice phones I have found several ways to do the second thing. http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers It works great. Jason Tom Vile wrote: I tend to stay away from the Grandstream phones for business use because they simply break to

[asterisk-users] Polycom Managment tools

Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config individually. Thanks

[asterisk-users] Cisco 7960 password/shared secret problem --- Related to OS X ?

Hello, Whenever I put in a password/Shared Secret in my 7960 and try and get it to register with asterisk on OS X setup, the phone fails to register. Oct 31 20:03:46 NOTICE[989]: chan_sip.c:11045 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for

[asterisk-users] Still no CLI in 1.4 branch (OSX)

I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?). I am currently

Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?

Dovid B wrote: Read the book Asterisk: The future of Telephony http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 It will teach you a lot. The trouble with this (I have it) is that it's dated. I do wish we had a more structured and maintained documentation project.

[asterisk-users] Re: Asterisk both behind a NAT and outside at the same time

BT == Brad Templeton [EMAIL PROTECTED] writes: BT The correct behaviour, as I see it is: BT a) Native bridge when connecting two external channels -- BT everybody is on the real internet b) Native bridge when connecting BT two internal channels -- everybody is on the 192.168.* network c) BT

[asterisk-users] Re: DTMF over IAX

On 2006-11-01 08:28:28 -0800, Jason Walker [EMAIL PROTECTED] said: Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the

Re: [asterisk-users] FXO Cards vs. Channel bank with T1

On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote: This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an FXO/FXS card. The issue here is an implementation bug of Zaptel

Re: [asterisk-users] Re: 1.4 branch on OSX?

Martin Joseph wrote: Good news! I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems to have resolved the asterisk hogging the whole CPU issue. I still can't use the regular console though (asterisk -r) as that is unresponsive. Using asterisk -c to start it , works and

Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

Thanks for the suggestions.. there is no such document in 1.2.6 in docs. On 11/1/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Matt wrote: Hi, I want to upgrade my older switch from 1.0.9 to 1.2.6 asterisk version. What do I need to be aware of? I AM aware 1.2.6 is not the newest

Re: [asterisk-users] Re: Opinions on the best wholesale origination/term providers

I wanted to buy service from SellVoip, however I have NEVER been able to reach anyone via phone, and I never really got email responses from them either.I have recommened a few times ISPhone ( www.isphone.net) however they don't have nationwide DIDs.On 11/1/06, Brad Templeton [EMAIL PROTECTED]

[asterisk-users] Java WEB Phone

Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar ms rpido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

Hello, The problem was wrong contexts defined like Marco said, and is solved. Now, i have another problem...of course :) On incoming calls, i only can receive calls if i define a line like the following, in extensions.conf: exten = _.,n,Dial(SIP/500,30,tr) (all incoming calls are redirected to

[asterisk-users] Remote-Party-Id and Attended Transfers

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and

RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?

Hi Andrew, I can highly recommend using the Granstream GXP 2000. Upgrade the firmware to ver. 1.1.1.14 and you won't have any problems. The 4 line buttons are not actual lines they are calls queued up on an extension so you can have as many incoming lines as you want. The first call comes in on

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

I strongly recommend you upgarde to the latest firmware for the GXP 2000. I have been using them for almost a year now and while the early firmware was poor they are now very stable and working fine (from 1.1.1.9) onwards. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada

Re: [asterisk-users] a2billing

Khaled wrote: Dear How can I customize a2billing to have two groups One have service to play its balance and the second group do not play the balance. This is not the a2billing support forum. Jeremy McNamara ___ --Bandwidth and Colocation

[asterisk-users] imap on debian

Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental: http://packages.debian.org/experimental/mail/uw-imapd On my test Etch system I simply downloaded the

Re: [asterisk-users] Still no CLI in 1.4 branch (OSX)

Martin Joseph wrote: I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is non responsive to commands (exit works though?).

Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?

On Wed, Nov 01, 2006 at 11:15:23AM -0700, Stephen Bosch wrote: Dovid B wrote: Read the book Asterisk: The future of Telephony http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 It will teach you a lot. The trouble with this (I have it) is that it's dated. I do wish we

Re: [asterisk-users] FXO Cards vs. Channel bank with T1

On Wed, Nov 01, 2006 at 08:29:32PM +0200, Tzafrir Cohen wrote: On Wed, Nov 01, 2006 at 09:08:43AM -0600, Ejay Hire wrote: This is incorrect. The data is still packetized and passed through IP which provides the same echo cancellation and distortion issues as a call that passed through an

Re: [asterisk-users] Polycom Managment tools

Neider, Clint wrote: Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config

Re: [asterisk-users] Re: Asterisk both behind a NAT and outside at the same time

On Wed, Nov 01, 2006 at 06:16:25PM +0100, Benny Amorsen wrote: BT == Brad Templeton [EMAIL PROTECTED] writes: BT The correct behaviour, as I see it is: BT a) Native bridge when connecting two external channels -- BT everybody is on the real internet b) Native bridge when connecting

Re: [asterisk-users] Polycom Managment tools

Neider, Clint wrote: Does anyone have a management tool for Polycom phones? For instance something to view software and boot versions of all the phones? I am looking for a product to remotely mange all phones in the environment without having to connect to each phones web config

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

I think I will agree with folks here, it must be something else on the network, not the phones themselves. I am not going to replace all of the phones, its too expensive, but for trial, want to try something better. PoE is also important to me at this point. I am thinking of trying Linksys 942. I

Re: [asterisk-users] Opinions on the best wholesale origination/term providers

Hi Brad,I can confirm the service quality of unlimitel.Have you look at www.les.net they provide both US and Canada DID. I heard good feedback about them On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote:I've been losing patience with my current provider, a small company called Sellvoip.Their

Re: [asterisk-users] DTMF over IAX

Jason Walker wrote: Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of

[asterisk-users] Remote-Party-Id and Attended Transfers

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and

Re: [asterisk-users] Upgrading from 1.0.9 to 1.2.6

Sorry, the file is located here: [EMAIL PROTECTED] ~]# ls -l asterisk-1.2.6/UPGRADE.txt -rw-r--r-- 1 1000 1000 8739 Dec 1 2005 asterisk-1.2.6/UPGRADE.txt Matt wrote: Thanks for the suggestions.. there is no such document in 1.2.6 in docs. On 11/1/06, Eric ManxPower Wieling [EMAIL

Re: [asterisk-users] Registration problem

Sergio R. D'Ippolito wrote: Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: */SIP/2.0 401 Unauthorized/* /Via: SIP/2.0/UDP

Re: [asterisk-users] Remote-Party-Id and Attended Transfers

Douglas Garstang wrote: Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party

[asterisk-users] Re: 1.4 branch on OSX?

On 2006-11-01 10:42:12 -0800, Joshua Colp [EMAIL PROTECTED] said: Martin Joseph wrote: Good news! I did an SVN update to my 1.4 branch again today, and 1.4-r46154 seems to have resolved the asterisk hogging the whole CPU issue. I still can't use the regular console though (asterisk -r)

[asterisk-users] Re: Still no CLI in 1.4 branch (OSX)

On 2006-11-01 09:09:26 -0800, Martin Joseph [EMAIL PROTECTED] said: I am testing 1.4 branch on OSX (10.4.8) and although it's running and passing calls ok, I am still not able to connect using asterisk -r. When I do open a CLI using asterisk -r, it appears to start up normally, but then is

[asterisk-users] Java Web Phone

Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance! Visita www.tutopia.com y comienza a navegar ms rpido en Internet.Tutopia es Internet para todos. ___ --Bandwidth and Colocation

Re: [asterisk-users] Diva server 4bri and Portuguese BRI

On Wed, 1 Nov 2006, Pedro Silva wrote: Hello, The problem was wrong contexts defined like Marco said, and is solved. Now, i have another problem...of course :) On incoming calls, i only can receive calls if i define a line like the following, in extensions.conf: exten =

Re: [asterisk-users] Which IP phones have best voice quality, preferably under $150

All the phones already have the latest firmware. They keep updating themselves automatically. In my setup of Grandstream phones, all the computers of the network go through the phones, i.e. I am using the builtin phones as swithces. They all have 2 ethernet ports. Does this has to do anything

[asterisk-users] Can I use Realtime entries to do multiple registers to same trunk/peer

I have a config where I define a single peer and have possibly hundreds of register commands for that single peer.I'm not clear if I can do the register part via Asterisk Realtime (right now I updated a file and force a reload which re-registers all the users defined in the register directives). I

RE: [asterisk-users] TE110P Card Little help

Hi, I really need some assistance in installing and configuring this card. I have already physically installed it into the computer which is running Mandriva 2006. I have compiled and installed asterisk 1.2.13 along with zaptel-1.2.10 and libpri-1.2.4. However I do not know what the next step

Re: [asterisk-users] imap on debian

On 21:47, Wed 01 Nov 06, Tzafrir Cohen wrote: Any potential testers eager to build imap storage support using proper Debian packages: Resonably up-to-date packages of c-client (uw-imap) 2004/2006 are by now only availble from experimental:

RE: [asterisk-users] Opinions on the best wholesale origination/termproviders

I am testing toll free and US DID inbound as well as A-Z outbound with les.net at the moment. Both the quality and support are quite good. Ping time to Vancouver is around 80ms. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcel Eric Loiselle Sent: Wednesday,

[asterisk-users] SMS and 1.2.12

Can anyone confirm that SMS() works correctly under asterisk 1.2.12? It used to work around version 1.2.7, but a few people have reported that 1.2.8 and 1.2.9 were a bit dodgy and that all their problems went away when they used app_sms from 1.2.7 in the later versions of asterisk. When a fixed

[asterisk-users] +Ura +md3200 nao encaminha ligacao

Salve Salve Galera. Tenho a seguinte situacao: Uma placa MD3200 ligada em uma linha telefonica comum(PTSN) e funcionando belezinha... Tenho configurado um URA, onde ele atende a ligacao que chegou no canal e solicita o numero do ramal de destino da ligação: Acontece que ao discar o ramal de

Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

Seems to me that you have a routing problem, asterisk should not know how to send packets to an outside IP using the NATed network. Make sure that the internal (NAT) interface doesn't have a gateway to it. On 10/31/06, Brad Templeton [EMAIL PROTECTED] wrote: I've read a lot of the descriptions

Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

Sorry for my previous post I misunderstood the problem. You should set canreinvite=no to all sip peers that connect from outside. On 10/31/06, C F [EMAIL PROTECTED] wrote: Seems to me that you have a routing problem, asterisk should not know how to send packets to an outside IP using the NATed

[asterisk-users] connecting internal line with external line

Hi, I'm new to asterisk. I want asterisk to connect a external line with an internal line: the PC dials a number and connects this call to a internal telephone (telephone switchboard, based on ISDN, 4 analogue telephones) of my office. Can somebody here give me keyword how to search (e.g.

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