[Asterisk-Users] SIP peer vs. user-- how is the USER ever selected?

Here's a real simple question for the Asterisk Venerable and Wise Ones: Help me understand how to name my section for the SIP user, so there is any hope of it ever being used in my sip.conf file. The Wiki says that it tries to match the user name from the From: header in the INVITE packet. If

Re: [Asterisk-Users] send SMS via own SMS Service

I've used Kannel (www.kannel.org) for long time; it works out Asterisk, but it's a good solution sometimes Use de latest CVS version! Best regards hi list, does anyone know how to configure asterisk to be able sending and receiving SMS over my own SMS gateway? it is

Re: [Asterisk-Users] chan_capi AVM C2

context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8

Re: [Asterisk-Users] chan_capi AVM C2

On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general]

[Asterisk-Users] capi.conf - AVM C4 P2P or P2MP

Quick question, I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI installed). Any help is greatly appreciated. -- Best regards Peer

[Asterisk-Users] PSTN gateway Asterisk - Virtual Switchboard???

Hi all, My imaginary scenario is the following one: I have a PSTN gateway called TotalControl1000, and I want to know, if connecting it to a visible server with asterisk with public IP I could config it to offer customers services of virtual switchboard. The customer would save the cost of a

Re: [Asterisk-Users] chan_capi AVM C2

Armin Schindler wrote: On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if i do that (with or without [interfaces]): [general]

Re: [Asterisk-Users] TE410P and SPANDSP

Hi, I also experienced broken page receiving fax on asterisk + spandsp with Digium TE410P. I also tried diff. versions of spandsp and asterisk, still no luck. I had no issues using the same asterisk + spandsp config with TE110P. Any ideas? At 09:21 AM 11/24/2005, you wrote: Hi, All

[Asterisk-Users] Join when empty problem, in queue

Hi all, when calling to a queue that has no agents logged in we expect to hang up, here is the extensions.conf queue configuration. exten= 2020,1,Answer exten= 2020,2,Ringing exten= 2020,3,Wait(2) exten= 2020,4,Queue(gestoria) exten= 2020,5,Hangup But althougth there isn't any agent it let us

Re: [Asterisk-Users] chan_capi AVM C2

On Wed, 14 Dec 2005, stéphane plichon wrote: Armin Schindler wrote: On Wed, 14 Dec 2005, stéphane plichon wrote: context=capi-in devices=2 This is just one section which two sets of options. You need to define two sections with [...]. See README. Armin no in or out call if

Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP

On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote: Quick question, I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn Putting P2P works, but I think is wrong. P2MP does not work (CAPI modules load, but capiinfo says no CAPI

Re: [Asterisk-Users] TE410P and SPANDSP

TE405p and spandsp works good. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: extension seen as busy when it is not

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail

[Asterisk-Users] Wildcard TDM2400P: comments

Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard TDM2400P. Does anybody have any comments with using this card, with any version of Asterisk, (maybe ill make this one Asterisk 1.2.x). I have had some stabilty issues using the 4 TDM400P. What

Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123

GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you also need to change GK

Re: [Asterisk-Users] Wildcard TDM2400P: comments

Can you define a LOT of pots line? Have you considered a channel bank. Here I'm running an ADTRAN 750. It's painless. You just need 1 T1 interface card for 24 lines. Jacques yusuf wrote: Hi all, we have the need for alot of plain analog lines. We thinking of buying the new Wildcard

Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP

Hello Armin, thanks for the quick response. I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn isdnmode=msn is in /etc/asterisk/capi.conf, but what about the /etc/isdn/capi.conf --- the configuration file for the capi modules?

[Asterisk-Users] voicemail boxes

HI I am new to asterisk. Could anyone pls tell me how do i create voicemail boxes Best Regards Vinod Yahoo! Shopping Find Great Deals on Holiday Gifts at Yahoo! Shopping ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Dial multiple destinations

Hey, When users call my phone at the office, I also want my mobile to ring.. This works fine using dial(..), but there may be some problems with the cdr's generated from this. There is only one cdr generated (for the first destination). I need to see if the call is answered by the

Re: [Asterisk-Users] voicemail boxes

After installing Asterisk, the first thing you'll need to do is add an extension. In the process of adding the extension, you can activate whether you want that extension to have voicemail or not. Have Fun Dakota - Original Message - From: Vinod To:

Re: [Asterisk-Users] Join when empty problem, in queue

Hi, try: joinempty = strict Morten - Original Message - From: Xavier Gil [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 10:19 AM Subject: [Asterisk-Users] Join when empty problem, in queue Hi all, when calling to a queue that has no agents

Re: [Asterisk-Users] Re: [helpp] Problem in astersik

Hi Guys After your guies replies now i have changed the machine .But this time i get little different problem i made following chnages in sip.conf [901] context=fromsip type=friend username=901 secret=901 callerid=Test2 901 host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw

[Fwd: Re: [Asterisk-Users] Re: [helpp] Problem in astersik]

---BeginMessage--- Hi Guys After your guies replies now i have changed the machine .But this time i get little different problem i made following chnages in sip.conf [901] context=fromsip type=friend username=901 secret=901 callerid=Test2 901 host=dynamic nat=yes canreinvite=no disallow=all

Re: [Asterisk-Users] voicemail boxes

Hi But that does not create the voice mail boxes. Is there any script which does it as mentioned in this link below http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 Regards VinodDakota [EMAIL PROTECTED] wrote: After installing Asterisk, the first thing you'll need to do

Re: [Asterisk-Users] capi.conf - AVM C4 P2P or P2MP

On Wed, 14 Dec 2005, Peer Oliver Schmidt wrote: Hello Armin, thanks for the quick response. I have an AVM C4 connected to a Mehrgeräteanschluss. What should I put into the /etc/isdn/capi.conf? isdnmode=msn isdnmode=msn is in /etc/asterisk/capi.conf, but what about the

Re: [Asterisk-Users] Dial multiple destinations

well, guess that's the way it is(a feature? i don't know)... but you can help ourself with the cmd SetCDRUserField respectively AppendCDRUserField (see http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCDRUserField) for dialing multiple destinations, maybe follow_me could be an interesting

[Asterisk-Users] [help] problem in astersik

Hi Guys After your guies replies now i have changed the machine .But this time i get little different problem i made following chnages in sip.conf [901] context=fromsip type=friend username=901 secret=901 callerid=Test2 901 host=dynamic nat=yes canreinvite=no disallow=all allow=ulaw

[Asterisk-Users] Help:asterisk 1.2.1 release compile

Hi all, I downloaded version asterisk 1.2.1 on my Debian. When i did make, i met this error: build_tools/make_version_hinclude/asterisk/version.h.tmp /bin/sh: line 1: build_tools/make_version_h: permisson non accordee make: ** [include/asterisk/version.h] erreur 126. It's not an error of

Re: [Asterisk-Users] Dial multiple destinations

Thanks, This sounds interesting, but it may cause some delay before it rings on my cell..? It is possible to implement a ringall strategy in the patch? Morten - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 14, 2005 12:31

[Asterisk-Users] quadbri, isnd, netherlands: callerid not working

hello, We installed an asterisk box last weekend dealing with 2 incoming groups of ISDN lines, and some 15 polycom phones. Works quite OK so far. But we have some problems that somehow I cannot resolve so far :-( The most urging issue is getting callerID to work. When I check logfiles, I see

[Asterisk-Users] Unable to find key

Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] -- Alejandro Vargas

[Asterisk-Users] subscription

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Re: [Asterisk-Users] extension seen as busy when it is not

Every few days our receptionist's phone will not take calls on one of the extensions. We have an extension 118 going to the first two lines of her phone and extension 101 going to the other. If we try to dial 118 it goes to voicemail even though she is not on the phone. Asterisk is

[Asterisk-Users] Exceptionally long queue in SIP Channel

Hi, Started getting a bombardment of these messages on the Asterisk console this morning (20+ a second): Dec 14 10:00:30 WARNING[17006]: channel.c:588 ast_queue_frame: Exceptionally long queue length queuing to SIP/bluecity29-a5cfDec 14 10:00:30 WARNING[17006]: channel.c:603 ast_queue_frame:

Re: [Asterisk-Users] SIP Subscription Storage Location

On 12/14/05, Brian Capouch [EMAIL PROTECTED] wrote: Douglas Garstang wrote: I can't understand why it was implemented this way (lack of design maybe?). Yep, that's it. Asterisk was designed by a bunch of fools who never even gave the first thought to what they were coding up. Yore

Re: [Asterisk-Users] Bonded ethernet ports and *

Rich - Even though I mentioned ethernet failover, I might have made it still a little too broad. The linux ethernet bonding module has been around for years, and there are several modes the linux bonding module can use which include: mode=0 (balance-rr) Round-robin policy: Transmit packets in

[Asterisk-Users] '#' (fast foward) and '*' (Rewind) not working in VoicemailMain

Hi, '#' (fast forward) and '*' (Rewind) not working in VoicemailMain with Asterisk 1.2.1 Do I have to do something in the dialplan to make this work? I have '##' set as a blind transfer and '*2' set as a attended transfer in features.conf. Per the Wiki Voicemailmain has the following

Re: [Asterisk-Users] Unable to find key

I think this is normal if you don't have call-forward-busy enabled. They key is deleted when it is disabled and added when enabled. - James Alejandro Vargas wrote: Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]'

Re: [Asterisk-Users] voicemail boxes

Vinod wrote: Is there any script which does it as mentioned in this link below http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=3 A script is no longer needed. Just edit your voicemail.conf file and add a line for each voicemail box. The first time someone drops into

RE: [Asterisk-Users] MGCP Unable to find key

Although I do not have an answer I changed the title so maybe someone with MGCP experience may notice it. Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable

[Asterisk-Users] fax and voice

Hello, I wish to configure Hylafax in order to send either fax or voice to Asterisk I've got a TDM400P (1FXS/1FXO) . What' s the best way to check the line to send fax or voice for incoming or outgoing ? Thanks for help H.G

RE: [Asterisk-Users] voicemail boxes

-Original Message- From: Vinod [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 14, 2005 5:26 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voicemail boxes HI I am new to asterisk. Could anyone pls tell me how do i create voicemail boxes Best Regards

[Asterisk-Users] Need help with sipura

I need helphow to config sipura 3000send and receive calls please. Thanks-- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Need help with Sipura 3000

Hello all, Please help me with the sipura 3000 how the Asterisk config need send and receive calls from Sipura 3000 What is Asterisk config need to input Thanks [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] Video calls (MS Messenger, Tandberg)

Hi, According to http://www.voip-info.org/wiki-Asterisk+video it should be possible to place video calls using asterisk. So far I managed to get both Microsoft Messenger and a video conference system from Tandberg to register with asterisk. Voice calls between both stations work perfectly (using

RE: [Asterisk-Users] Hint Priority for Polycom Phones

Title: Re: [Asterisk-Users] Hint Priority for Polycom Phones Hello Doug, I assume you have subscribecontext set in sip.conf right? Also,I have a 601/sidecar and have the hints working fine on the first registration. On my second server registration they are not yet coming through. I am not

[Asterisk-Users] appradius

Hello all. Has anybody works with appradius? where can I find documentation? Regards, Jsalas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Asterisk as client to PortaBilling

I am trying to register my Asterisk to a Portabilling system. My asterisk registers with no problems, but when i try to send calls to portabilling I get the response: Dec 14 15:23:31 WARNING[420]: chan_sip.c:727 retrans_pkt: Maximum retries exceed ed on call [EMAIL PROTECTED] for seqno 104

[Asterisk-Users] Re: Testing 10.0.0.203 with 10.0.0.0

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... FC4, Asterisk 1.0.9 and SjPhone softphone. On CLI I get this message every 20 sec. # Testing 10.0.0.203 with 10.0.0.0 10.0.0.203 is the IP of softphone and 10.0.0.0 is the network defind in sip.conf. Asterisk server is on

Re: [Asterisk-Users] Partial PRI pass thru?

I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any. Unfortunately I cant speak for the 2nd gen. cards as we haven't used them. The 1st

Re: [Asterisk-Users] Bonded ethernet ports and *

Rich - Even though I mentioned ethernet failover, I might have made it still a little too broad. The linux ethernet bonding module has been around for years, and there are several modes the linux bonding module can use which include: mode=0 (balance-rr) Round-robin policy: Transmit packets

Re: [Asterisk-Users] Need help with sipura

I need help how to config sipura 3000 send and receive calls please. Go to www.voxilla.com and run their config wizard. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.

Hi, Please excuse the double post but I am about to report this as a bug and I want to verify that others are having the same problem. Also I have seen numerous bugs reported that are not bugs but just misconfiguration, etc. and I do not want to burden the developers with a frivolus bug

RE: [Asterisk-Users] Partial PRI pass thru?

\ I'd recommend the Digium dual port cards - generation 2 card are excellent and the support we receive superb. And it supports Digiums support and development of Asterisk - Sangomas contribution is token if any. Unfortunately I cant speak for the 2nd gen. cards as we haven't used them.

Re: [Asterisk-Users] cdr_addon_mysql can't find libmysqlclient.so

Hi! Dec 13 12:19:29 WARNING[4112]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory Mine is in /usr/lib/libmysqlclient.so, so how about just adding a symlink? Philipp ___ --Bandwidth

Re: [Asterisk-Users] Unable to find key

Hi! Is this normal? Can I ignore this messages? Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' Dec 14 13:18:06 DEBUG[5312] db.c: Unable to find key 'MGCP/aaln/[EMAIL PROTECTED]' in family 'cfb' [...] Look like you have enabled

[Asterisk-Users] Best way to automatically stop and start Asterisk nightly

Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak (Asterisk slows down over time). I have reviewed the Asterisk suggestions for management and one item is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is the recommend

[Asterisk-Users] Gateway crashes when transferring to external lines

I'm using an Asterisk system with a Zultys MX250 as a media gateway for our PSTN. When I configured this originally, I couldn't make or receive any outside calls - the MX250 would actually crash and restart itself. Very bad. As I was troubleshooting, I found a tip in the Asterisk wiki that

Re: [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.

OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same problem with 1.0.9 and 1.2.0 Chuck Bunn wrote: Hi, Please excuse the double post but I am about to report this as a bug and I want to verify that others are having the same problem. Also I have seen numerous bugs

Re: [Asterisk-Users] IP Phone Recommendation

Kris, I highly recommend the snom 320. Very easy to configure, and very easy to setup line appearances. As has already been mentioned, the idea of lines is a bit dated. For more information, read this: http://forums.digium.com/viewtopic.php?t=891. Sean Duracom ISP Lists wrote: We

[Asterisk-Users] hardware echo cancellation for TDM card

Hi Just checking, Is there any hardware echo cancellation card available for the digium TDM400P card I read the archives and could not find any. I think I need the TDM2400 card for this Thanks Patrick ___ --Bandwidth and Colocation provided by

RE: [Asterisk-Users] RE: Asterisk to Avaya IP Office

Title: Re: [Asterisk-Users] TDM01B vs. X100P I fixed this by making the extensions on the asterisk box DIDs . So when the IPO passed the 3 digit extension number then the asterisk box looked at that and sent it on to the extension. Asterisk seemed to see all H323 calls as outside calls

RE: [Asterisk-Users] Join when empty problem, in queue

in queues.conf change joinempty = no leavewhenempty = no to joinempty = strict leavewhenempty = strict From: Xavier Gil [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject:

[Asterisk-Users] T1 to T1 dialout problem

I need a few minutes of time to work out a dial out problem. I'm willing to pay for your time. What I have is a system that connect 2 external VMS systems to one of two Telco T1's. Mainly the Telco T1's route inbound calls to one of the two external VM systems depending on the DNIS. This

Re: [Asterisk-Users] WIFI Phones

rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip

[Asterisk-Users] OT: Linux on treo 650

http://www.engadget.com/entry/1234000497072377/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Need help with sipura

Really? Did you try reading classes? On 12/14/05, Talkvoip Telecom Canada [EMAIL PROTECTED] wrote: I need help how to config sipura 3000 send and receive calls please. Thanks -- [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] hardware echo cancellation for TDM card

On 12/14/05, Patrick Fortin [EMAIL PROTECTED] wrote: Hi Just checking, Is there any hardware echo cancellation card available for the digium TDM400P card I read the archives and could not find any. I think I need the TDM2400 card for this No. Not at this time. You will need indeed

Re: [Asterisk-Users] hardware echo cancellation for TDM card

Patrick Fortin wrote: Is there any hardware echo cancellation card available for the digium TDM400P card No. Software echo cancellation is fine for small density applications like 4-8 ports, unless you are using a very low performance CPU. ___

Re: [Asterisk-Users] Polycom 501 remapping keys

What I really want to be able todo is use the services button or any of the other buttons that serve no purpose right now. I would like to have it start a page (which on my * box is just dialing a particular extension), I have this working on my Polycom 501's using the 3rd line appearance,

[Asterisk-Users] Cisco 7940 Time Source

Does anyone have an idea of where cisco 7940's get their time from? Up until monday (when our dns server crapped out so we killed it), our phones all had time... now they only show time when they're just rebooted and it's only for a few minutes. Any ideas? Aaron Daniel

[Asterisk-Users] Headset Phones?

I am looking for IP phones that are headset and then a phone that clips on your belt. I have been looking at wi-fi phones, but I'm not sure about headset capabilities. I'm using these with *, so compatibility with * is required. Can anyone offer any suggestions? ~kurth -- Kurth Bemis

Re: [Asterisk-Users] WIFI Phones

combine an iaxy and a plain old cordless phone. It works great for me around the workplace. I think there are even four-line, five-handset cordless systems, although you'd need four iaxys as well. Moj rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them?

Re: [Asterisk-Users] WIFI Phones

On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote: rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :) not entirely true if you expand your definition :P I have an ipaq which is capable of acting like a soft phone (and it does,

[Asterisk-Users] traffic shaping

Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth reservation policy in *. What about using * with 2 network cards betwen the LAN and ADSL router

[Asterisk-Users] Background() followed by Read - something wrong?

Hi, I'm using Asterisk 1.2.1, and have been trying to sue the Background() command followed by Read() (for a screening app, but that's beside the point) I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) ...

Re: [Asterisk-Users] voicemail boxes

Dakota wrote: After installing Asterisk, the first thing you'll need to do is add an extension. In the process of adding the extension, you can activate whether you want that extension to have voicemail or not. Have Fun Dakota Sounds like someone is using [EMAIL PROTECTED] (or at least

[Asterisk-Users] sharing a line w/multiple extensions

I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP extension phone can pick up and join the conversation. How do I configure this in extensions.conf?

Re: [Asterisk-Users] Cisco 7940 Time Source

If you change your config for your phones to use unicast for your SNTP Mode, the time should stay. I had the same problems until I changed it to unicast.On 12/14/05, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone have an idea of where cisco 7940's get their time from?Upuntil monday (when our

Re: [Asterisk-Users] g729 translation to zap (ISDN) doesn´t work

I thougt i have some problems with ztdummy and removed that # in front of ztdummy in the zaptel Makefile before compiling. But still no change. I even tried it with another Phone, a Planet VIP-150T. Still the same Problem, i don´t hear anything from the SIP Phone on the ISDN Phone, but i hear

RE: [Asterisk-Users] Headset Phones?

Kurth: The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000 also has a jack for headset/ ear bud use. Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kurth

Re: [Asterisk-Users] WIFI Phones

and that ipaq can do iax2 ?? Guess not cheers klaus trixter aka Bret McDanel schrieb: On Wed, 2005-12-14 at 17:33 +0100, Matt Riddell wrote: rossi.tek wrote: I'm looking for iax2 wifi phones, do you know where i can buy them? Yes. Nowhere. :)

[Asterisk-Users] asterisk + H323 + 723

Hi I am using asterisk 1.2.1, does any one has any luck with asterisk and h323. I want to use the codecs 723 and 729 with it. I am having one way audio issues with oh323 with I receive a call to asterieks through 723 . is there a successful implementation ? regards kani

RE: [Asterisk-Users] traffic shaping

http://www.krisk.org/astlinux/misc/astshape kicks butt hth -Original Message- From: Jose Limeres [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 14, 2005 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] traffic shaping

Re: [Asterisk-Users] sharing a line w/multiple extensions

Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst multiple extensions. i.e. If one SIP phone answers the call, another SIP

Re: {Scanned} Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123

Alessio Focardi wrote: GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you

Re: [Asterisk-Users] sharing a line w/multiple extensions

Let me revise this a little: I'd like to configure Asterisk so an incoming call from one POTS line is shared amongst multiple extensions - both SIP and analog. i.e. If one SIP phone answers the call, another SIP or analog extension phone can pick up and join the conversation. How do I

Re: [Asterisk-Users] traffic shaping

look into linux advanced routing and traffic control lartc Casey Boone Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using traffic shaping embeded with *? As in our case it is not possible to configure it in the ADSL router we would like to implement some kind of bandwidth

Re: [Asterisk-Users] traffic shaping

Jose, I don't know what everyone else uses, but I use wondershaper. It's a bit rough, but it does the job well for what I need ( prioritize voip traffic over everything else ). Google should bring it up for you. Sean Jose Limeres wrote: Hi all, Has anyone a good piece of advice on using

Re: [Asterisk-Users] Headset Phones?

Maybe a bit simplistic... ATA with any cordless phone and walmart headset... Garrett Smith wrote: Kurth: The UTStarcom F1000 has an ear bud for it. I believe the Hitachi IPF-5000 also has a jack for headset/ ear bud use. Garrett Smith [EMAIL PROTECTED] 716-250-3408 Direct 716-903-9495 Cell

RE: [Asterisk-Users] OT: Linux on treo 650

But can they run Asterisk and create an IAX trunk back to your PBX while running a softphone? I thought not. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, December 14, 2005 8:44 AM To: Asterisk Users Mailing List -

[Asterisk-Users] HOWOT transfer call from mobile back to extension?

Cheers all. I hope I did not miss this in my quick searching for this, and if I did, apologies, please just a minor scolding as you point me at the URL. But, if I did not miss it, then is there any one out there who has figured-out how to skin this cat. I have a few people's mobile phone

Re: [Asterisk-Users] sharing a line w/multiple extensions

also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) C F wrote: Look at meetme, also FOP (www.asternic.org) can do that for you. On 12/14/05, Robert La Ferla [EMAIL PROTECTED] wrote: I'd like to configure Asterisk so that incoming calls from one POTS line are shared amongst

Re: [Asterisk-Users] sharing a line w/multiple extensions

Sean Cook wrote: also you can ring multiple extensions: Dial(SIP/101SIP/102SIP/103) I have that but once one extension picks up, others can't join in. Well, at least when I tried it with mixed SIP and Zap, it didn't work. Maybe all SIP does but I need it to work for all phones SIP and

Re: [Asterisk-Users] Background() followed by Read - something wrong?

I did the following s,1,Background(blablabla) s,2,Read(VARIABLE||1) ; accepting only one digit (1 to accept call, anything else to hangup) That's not the right approach. Do something like his: [confirmcall] exten = s,1,Background(blablabla) exten = 1,1,Goto(accept_call_context,s,1) exten =

RE: [Asterisk-Users] traffic shaping

a simple m0n0wall or pfsense system running on a sub $100 pc makes a GREAT router for this and allows you to use multiple internet connections in concurrency for speed increases other than that, yes, 2 NICs and some creative networking and you're done From: [EMAIL PROTECTED]

[Asterisk-Users] ChanIsAvail() and SIP

Hi everyone, I have started trying to use ChanIsAvail() to detect when a phone is in use (on any call) and my results are disappointing. Here are some examples out output to the console followed by the meaning of the return status code based on what I have found in the comments on this

[Asterisk-Users] 1.2.1 Compile Error

I'm trying to compile 1.2.1 for the first time and I am getting a compile error that I can't figure out. I've compiled 1.0.x many times without this issue, although it has been on different boxes. The error is configure: error: termcap support not found, which is odd because when I do a rpm

[Asterisk-Users] Re: 1.2.1 Compile Error

Oops, my bad. 5 minutes after I sent it, I realized I was missing ncurses-devel. Peder @ NetworkOblivion wrote: I'm trying to compile 1.2.1 for the first time and I am getting a compile error that I can't figure out. I've compiled 1.0.x many times without this issue, although it has been on

Re: [Asterisk-Users] HOWOT transfer call from mobile back to extension?

I use this at work. You have to make sure you use the right T / t options when dialing the mobile, then just use the standard # transfer. I changed ours to ##. - James [EMAIL PROTECTED] wrote: Cheers all. I hope I did not miss this in my quick searching for this, and if I did, apologies,

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