Re: [asterisk-users] Doorphone vs. Grandstream BT101

Jay Milk a écrit : Doug Lytle wrote: [...] Thanks to Dave and Doug for the quick responses. I'm looking forward to hearing the response on #3, but I think I'll get get one of these devices to play with this weekend. At worst, it'll be a usable garage or basement phone. Hi Jay, sorry for j

Re: [asterisk-users] Problem with ztdummy

On Mon, Mar 26, 2007 at 08:55:15AM +0100, Alan Chandler wrote: > On Monday 26 March 2007 05:27, Tzafrir Cohen wrote: > > > You have the module zaptel from an older version still loaded . > > > > /etc/init.d/zaptel unload > > /etc/init.d/zaptel start > > That didn't fix it - but a reboot of the sy

[asterisk-users] Failure acknowledgement time

Hi, I've noticed that if I disconnect or reconnect a phone from the net, Asterisk take long time to realize that (even more then 10 minutes). Is there a way to reduce this time, working on the configuration files? Thank you. silvia -- Passa

Re: [asterisk-users] Failure acknowledgement time

You can use qualify=(time in ms) option in sip.conf but its phone's configuration that should register to asterisk everytime its reconnected . On 26/03/07, cimsi <[EMAIL PROTECTED]> wrote: Hi, I've noticed that if I disconnect or reconnect a phone from the net, Asterisk take long time to realiz

RE: [asterisk-users] Anyone having trouble with claling US Domesticon Sellvoip?

I opened up a ticket with them, but I'm not holding my breath. I think it's time to start moving my DID's before the inbound stops working. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: Sunday, March 25, 2007 9:18 PM To: Asterisk Users Mailing List - N

Re: [asterisk-users] How to resolve CallerID from AudioCodes FXO

Hi Hind, Sorry, I haven't work for it yet. I still left on the number of endpoints assigned. Probably I will concentrate on it once I finish with my gnuDialer Project. I'll keep you informed. Angel hind habaoui <[EMAIL PROTECTED]> wrote: hi angel. it is about the CallerId, i have the same pro

[asterisk-users] Moving from Bristuff to mISDN

Hi, Which benefit-features did you gain-loose when moving from Bristuff to mISDN ? Would you recommend such a move ? Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Failure acknowledgement time

Could you give me an example of how to configure this parameter in sip.conf? How does it work with the maxexpiry and minexpiry parameters? Thank you! -- Initial Header --- >From : [EMAIL PROTECTED] To : "Asterisk Users Mailing List - Non-Commercial Discussion" ast

Re: [asterisk-users] voicemail is not playing messages

Joseph wrote: That is the exactly the same permission on my asterisk-1.2.13 that is working OK. I concur with Tzafrir, check the permissions of the directory itself. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve ne

[asterisk-users] cutting hash in dial app

hello, isit possible to cut off the hash behind a dial string? coz we have a provider who gives us an error 600 "Declined" if ther is a hash in dial command. for example: Dial("SIP/x.x.x.x-b7d2d870", "SIP/[EMAIL PROTECTED] x") and i have to cut out: -b7d2d870 regar

Re: [asterisk-users] Doorphone vs. Grandstream BT101

marcotasto a écrit : I did something similar one year ago for a friend [...] If you are interested, I can post my results and the link to my site when they will be ready. Yes please, would be great. Many thanks :-) -- Daniel ___ --Bandwidth and C

Re: [asterisk-users] Limit call duration

Rizwan Hisham wrote: I think you can set absolute timeout variable for incoming call also. I havent tested it yet, y dont you try it. do like this: before every local extension you can set: exten=> _XXX,1,SET(Timeout(absolute) = 10) exten=> 123,2,Dial exten=> 234,2,Dial > Yuan Liu

Re: [asterisk-users] Moving from Bristuff to mISDN

Hi Olivier, ISDN channels are not seen as zap channels as with briStuff so you have to edit misdn.conf to use them. If you are using your own framework above Asterisk architecture you to make some changes to manage new type of channels. Giorgio Incantalupo Olivier wrote: Hi, Which benefi

[asterisk-users] Counting callers

Hi, Can I count and say back to caller how many calls waiting on current extension? -- Suich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/ma

Re: [asterisk-users] how to check and set D-channel status

Hi, can you explain so more :) ___ --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: Re: [asterisk-users] Problem with busy and unavailable

Hello James, Am Freitag, 23. März 2007 21:23 schrieben Sie: > You might be reading out of date documentation. In 1.2.14, if you don't > have "priorityjumping=yes" in extensions.conf and you don't activate > priority jumping for an individual application (for Dial(), you add "j" to > the options s

Re: [asterisk-users] Moving from Bristuff to mISDN

Hi, Beside having to use misdn.conf instead of zaptel.conf, did you notice any gain or lost moving from bristuff to misdn ? I was thinking about callerID, compliance to Telco ISDN, ... From my point of view, it's pretty delicate to wait for a new Junghanns version to be published but we're use

[asterisk-users] Asterisk incoming caller id problem

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4

Re: [asterisk-users] Asterisk incoming caller id problem

On Mon, 26 Mar 2007, johnny_xing wrote: Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller I

Re: [asterisk-users] Moving from Bristuff to mISDN

Hi Olivier, what would you like to have that junghanns driver does not give you? Giorgio Olivier wrote: Hi, Beside having to use misdn.conf instead of zaptel.conf, did you notice any gain or lost moving from bristuff to misdn ? I was thinking about callerID, compliance to Telco ISDN, ... F

RE: [asterisk-users] AsteriskNow Beta 4 with T1 Cards?

> Tzafrir Cohen wrote: > The asterisk-gui uses /etc/asterisk/zapscan.conf, which is detected > at boot time (by zapscan which is run from /etc/init.d/zaptel) > As genzaptelconf is now able to replace the zapscan utility of the > > asterisk-gui, and also to detect digital spans and give them > r

Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?

Can you tell me, why sellvoip rocks ? I am looking to sign up with someone. Thanks, balu raman On 3/25/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: Salvatore Giudice wrote: > Nothing has changed in my Asterisk configuration and now outbound US is > getting nothing, but 403's. Anyone else having

[asterisk-users] No Audio when integrating with openSER and Asterisk in the SAME LAN ,

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER

Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?

Balu, I suspect the author was expressing sarcasm. On 3/26/07, Balu Raman <[EMAIL PROTECTED]> wrote: Can you tell me, why sellvoip rocks ? I am looking to sign up with someone. Thanks, balu raman On 3/25/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: > Salvatore Giudice wrote: > > Nothing has ch

Re: [asterisk-users] Asterisk incoming caller id problem

On Mon, Mar 26, 2007 at 01:35:36PM +0100, Gordon Henderson wrote: > I've had issues with caller ID on TDM400 cards myself. I've never gotten > fully to the bottom of it either. Sometimes it works, and sometimes I get > nothing. (when a "normal" phone on the same line gets the ID OK all the > tim

[asterisk-users] ARI with * 1.4.x

Afternoon A little off-topic...but... Does any1 know why recorded call with "IAX2" in the filename are not displayed within ARI? LittleJohn's website isn't a helpful place for ARI (o; cheers rick ___ --Bandwidth and Colocation provided by Easynew

[asterisk-users] Re: Two or More Bri Cards

I Always had very bad experiences with 2 HFC cards in the same box I strongly suggest you to use a dual port card Regards Edoardo Farooq Ahmed ha scritto: hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especia

RE: [asterisk-users] Re: Two or More Bri Cards

Edoardo Serra wrote: > I Always had very bad experiences with 2 HFC cards in the same box > > I strongly suggest you to use a dual port card > > Regards > > Edoardo Interesting... I mean one would think that is the case all the time. In another words, that is logical, and I though the same but

[asterisk-users] How is this feature called ?

Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Best regards ___ --Bandwidth and Colocation provided by Ea

Re: [asterisk-users] Anyone having trouble with claling US Domestic on Sellvoip?

Tom Lynn wrote: > Balu, > I suspect the author was expressing sarcasm. Tom: Thank you for preventing a tragedy. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] 1.4 - IAX2 - No registration for peer

hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' rejected: 'Registration Refused' from: '10.10.10.82' I was getting a 'Cause Code: 29' INV,POKE,...,REJ but I can't duplicate that level of debugging agai

[asterisk-users] Registration timed out after a "sip reload"

Hi, I have configured a sip provider account, with register => user :[EMAIL PROTECTED]/user, with Asterisk 1.4.2. Then I start Asterisk, which register successfully to the sip provider: sip show registry show me the provider and status "Registered". I do a "sip reload" in the CLI, and now

Re: [asterisk-users] AsteriskNow Beta 4 with T1 Cards?

On Mon, Mar 26, 2007 at 09:18:25AM -0400, Brian K. Alexander, Jr. (Vision Point Systems) wrote: > > Tzafrir Cohen wrote: > > The asterisk-gui uses /etc/asterisk/zapscan.conf, which is detected > > at boot time (by zapscan which is run from /etc/init.d/zaptel) > > > As genzaptelconf is now able

[asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position

Hi Chris, Have you found a solution to this problem? I have been dealing with a similar issue on and off for a few months now. I get these format_wav errors out of the blue and the logs get rotated many times a second into oblivion. The only thing that is different on this box versus my othe

[asterisk-users] Asterisk and T38 ?

Hi i read the list and see a lot of personn say T38 it's not possible with asterisk and other says that he use T38 with asterisk ?? i don't understand ;=) He have a solution (commercial or free) to add T38 ? I have : Fax Machine --> Linksys PAPT --> Asterisk ===> IAX2 on Sdsl ===> Asterisk

Re: [asterisk-users] Counting callers

Do you mean queue? If so, yes this is a very easy thing to do and is document on the voip-info.org wiki under the queues section. On 3/26/07, Suity Zsolt <[EMAIL PROTECTED]> wrote: Hi, Can I count and say back to caller how many calls waiting on current extension? -- Suich _

Re: [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position

Bartek Bulzak wrote: Hi Chris, Have you found a solution to this problem? I have been dealing with a similar issue on and off for a few months now. I get these format_wav errors out of the blue and the logs get rotated many times a second into oblivion. The only thing that is different on t

Re: [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position

Once I updated asterisk it went away. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed

[asterisk-users] Multi-registration ?

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards

Re: [asterisk-users] Multi-registration ?

olivier, soft phones on a PC require a port to connect to the server... haven't tried multiple soft phones, simultaneously, connecting to one server or multiple servers but if you can configure the outgoing port, it should be possible... NAT might get quite confusing so I would try it before

[asterisk-users] outbound call

HI All, I am new to asterisk. i want to make outbound calls from asterisk. I tried with many times with the given settings but in vain This is my scenario: I have a *user A* who has registered with sip server(ONDO), I made asterisk to register as a sip client with ONDO, I want to make a call

Re: [asterisk-users] Re: format_wav.c:247 update_header: Unable to find our position

Eric "ManxPower" Wieling wrote: I have found that if the logs are rotating several times per second then the problem is either you are out of disk space, or you have too many files in 1 directory. When it happened to me I had massive numbers of files in /var/log/asterisk. FYI I have finally

Re: [asterisk-users] Re: Two or More Bri Cards

Interesting... I mean one would think that is the case all the time. In another words, that is logical, and I though the same but recently we have installed: 4 x one port and 1 X 4 port cards into a same box running PBXware. That is 5 cards in total... No complaints for 3 months running 2-3 thous

[asterisk-users] Polycom 601 loop

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. T

Fwd: [asterisk-users] Multi-registration ?

olivier, soft phones on a PC require a port to connect to the server... haven't tried multiple soft phones, simultaneously, connecting to one server or multiple servers but if you can configure the outgoing port, it should be possible... NAT might get quite confusing so I would try it before maki

[asterisk-users] Re: Polycom 601 loop

Sorry, forgot to attach the sip.conf and extensions.conf files. Attached now. [general] context=from-sip; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=actarg.com; Realm for digest authenticati

[asterisk-users] SER vs Asterisk?

Hi Marco - since you've asked. :) We're looking to have 3 to 4 T1 lines dropped into our datacenter, which will contain one of the Asterisk boxes. From the data center, AT&T is going to be running a fiberoptic connection directly to our callcenter, which will employ about 75-100 people. We'll h

RE: [asterisk-users] Re: Two or More Bri Cards

A lot of the problems people are having with ISDN have to do with the service provider setup and not the cards (provided that we are talking about 2-3 HFC cards) For example in Greece the local telephone company uses 3 diffrerent types of ISDN equipment in their centers (Siemens,Ericsson,Alcatel) W

Re: Fwd: [asterisk-users] Multi-registration ?

Olivier wrote: olivier, soft phones on a PC require a port to connect to the server... haven't tried multiple soft phones, simultaneously, connecting to one server or multiple servers but if you can configure the outgoing port, it should be possible... NAT might get

Re: [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC

What version of zaptel are you using fxotune from? You should be using 1.4 fxotune, between 1.2 and 1.4 there was a pretty significant rewrite that will probably give you more improvement. Also, just in case you hadn't, make sure you're doing an fxotune -s on bootup so that it applies you

Re: [asterisk-users] Asterisk incoming caller id problem

On Mon, 26 Mar 2007, Mike Jagdis wrote: On Mon, Mar 26, 2007 at 01:35:36PM +0100, Gordon Henderson wrote: I've had issues with caller ID on TDM400 cards myself. I've never gotten fully to the bottom of it either. Sometimes it works, and sometimes I get nothing. (when a "normal" phone on the sam

Re: [asterisk-users] SER vs Asterisk?

David, I have done a very successful implementation that is almost identical to what you are describing. Do you have in-house developers for the customization to your CRM? Would you just need guidance or would you need someone to dig into your system? I designed and implemented a very sim

RE: [asterisk-users] Re: Two or More Bri Cards

Hi Senad, Could you elaborate ? Which type BRI cards did you mix ? Which driver and channel ? -- Hi HFC based chip based one port cards and Jugnhaans quadBRI using bristuff drivers. We have spent log time making sure the bristuff drivers are handled correctly for our customers ne

[asterisk-users] Getting an ASR Number

Does anyone have any suggestions (existing scripts, etc) for figuring out ASR? We wanted to write something to be a little more in depth then what we have now. Wasn't sure if anyone had any suggestions for things that might be pre-existing, or what you've done to accomplish this. ___

[asterisk-users] cutting hash in dial app

<>___ --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: Fwd: [asterisk-users] Multi-registration ?

I tried using multiple accounts from one phone to separate call centre traffic but the phones (Aastra 480i) would default all calls from the phone to the account with the "highest" line number. This made it impractical for my purposes. regards, Drew Do think this limitation comes the phone

Re: [asterisk-users] Re: Two or More Bri Cards

On Mon, Mar 26, 2007 at 06:50:00PM +0100, Senad Jordanovic wrote: > HFC based chip based one port cards and Jugnhaans quadBRI using bristuff > drivers. Using the florz patch for zaphfc? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-79524

[asterisk-users] Hang up detection time in FXS module

Hello, I have one TDM400 card with two FXS ports and two standard analog telephones connected on them. When I put the phones on hook, the Asterisk delays about two seconds to recognize hang up. This delay is from TDM400, from Zaptel or from Asterisk? Can I decrease this time with any configurat

Re: Fwd: [asterisk-users] Multi-registration ?

Olivier wrote: I tried using multiple accounts from one phone to separate call centre traffic but the phones (Aastra 480i) would default all calls from the phone to the account with the "highest" line number. This made it impractical for my purposes. regards, Drew

Re: [asterisk-users] Asterisk incoming caller id problem

On Mon, Mar 26, 2007 at 06:19:15PM +0100, Gordon Henderson wrote: > That's less of an issue (for me) - I'm guessing that most UK analogue > phones with CID display will support *anything* that the telco throws at > them, probably as BT, Telewest/NTL/Virgin Media (?) all seem to be > different (a

Re: [asterisk-users] 1.4 - IAX2 - No registration for peer

Its been a long day, so I am probably missing something, but why are you 'qualify'ing a user entry? On 26 Mar 2007, at 16:36, dave cantera wrote: hi, I'm getting registration errors I can't debug... [Mar 23 11:07:20] NOTICE[2952]: chan_iax2.c:7344 socket_process: Registration of 'host2' reje

[asterisk-users] Doorphone

We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem is that we can't send DTMF signals to the door to

Re: [asterisk-users] Doorphone vs. Grandstream BT101

Steve Totaro wrote: Just get a Grandstream ATA and a handset with no buttons. So simple. That doesn't really meet my needs -- I want to be able to dial-out, and have the person on the other end simply be able to push a button to ring the doorbell. The doorbell button requirement stems from th

Re: [asterisk-users] Chan_cellphone and CentOS 4.x

Bruce Reeves wrote: I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellpho

Re: [asterisk-users] Doorphone

Ray Wadkins wrote: We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem is that we can't send DTMF signa

[asterisk-users] SIP registration

When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip;

Re: [asterisk-users] Polycom 601 loop

nathan, can you post your extensions.conf file [to-sip], and your sip.conf section for extension 201... ie [201]? it looks like, perhaps, it is a dialplan problem... daveC Nathan Bell wrote: I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can succes

[asterisk-users] Device not registering after boot

Hi folks. I'm having a problem with a SIP-enabled device that doesn't seem to want to register after it reboots. If I program the device manually via its interface, it registers just fine. However, once I reboot it, it fails to register with Asterisk, despite all the proper information being

Re: [asterisk-users] Polycom 601 loop

Pertinent part of extensions.conf: ; from outside T1 [from-ptsn] exten => s,1,Answer() include => cac-ext include => sip-ext include => intertel-ext exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() ; from sip lines [from-sip] include => internal ; generic interal route [internal] exten =>

Re: [asterisk-users] SER vs Asterisk?

David : will contact you offline as you requested. Just a general comment from your project description. You may do well to consider the amount of call center features and statistics you will require. Without knowing all I suspect that you will find the project will work best if the home based a

Re: [asterisk-users] Re: Polycom 601 loop

nathan, try dial() directly to the extension [to-sip] exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],120) try exten => _X.,1,Dial(SIP/${EXTEN},20) where ${EXTEN} = 201 and [201] in /etc/sip.conf is [201] type=friend; Friends place calls and receive calls context=from-sip

[asterisk-users] Emergency chan_sip issue

Greetings list, Wondering if some kind soul can help me with an issue with chan_sip segfaulting as soon as it loads... Basically, if sip.conf contains any peers with "host=dynamic" in them, asterisk won't start. Doing -vvvdddc yields the following: [chan_sip.so] => (Session Initiation Protocol

[asterisk-users] SRTP vs ZRTP in Asterisk

Hi All, I've been reading about Phil Zimmermann's ZRTP encryption scheme for SIP clients. This seems attactive but I don't use soft phones. I'm guessing that we'd need ZRTP support in Asterisk in order to use it to secure calls from hard phones. There seem to be issues with SRTP key exhange bet

Re: [asterisk-users] Re: Polycom 601 loop

No loop now, but instead I get this: Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Mar 26 15:42:18 DEBUG[1854] app_dial.c: Exitin

RE: [asterisk-users] Emergency chan_sip issue

If "something" changed on its own 6 hours ago (i.e. you didn't touch Asterisk or its config), then look beyond Asterisk. Did you do a hard reboot yet? Memory check? Reseat PCI cards and memory? Check power supply? New drivers/software loaded? (Did YUM run in the background?) If everything el

Re: [asterisk-users] Re: Polycom 601 loop

Hi Nathan - No loop now, but instead I get this: Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Mar 26 15:42:18 VERBOSE[1854] logger.c: == Everyone is busy/congested at this time (1:0/0/1) Mar 26 15:42:18 DEBUG[1854] app_dia

Re: [asterisk-users] Re: Polycom 601 loop

and sip show users Noah Miller wrote: Hi Nathan - No loop now, but instead I get this: Mar 26 15:42:18 NOTICE[1854] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Mar 26 15:42:18 VERBOSE[1854] logger.c:   == Everyone is busy/

Re: [asterisk-users] SIP registration

Hi Nathan - I just saw this post about having trouble registering your phone ;-) When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '' failed for '192.168.3.2' - Not a local SIP domain sip.conf

Re: [asterisk-users] Re: Polycom 601 loop

This is what I get from the asterisk CLI: ast*CLI> sip show peers Name/username HostDyn Nat ACL Port Status 202(Unspecified)D 0Unmonitored 201(Unspecified)D 0Unmonitored 2 si

Re: [asterisk-users] TDM analog cards, volume, echo, fxotune, ztmonitor and HPEC

Hi, Matt: Matthew Fredrickson wrote: > What version of zaptel are you using fxotune from? You should be using > 1.4 fxotune, between 1.2 and 1.4 there was a pretty significant rewrite > that will probably give you more improvement. Also, just in case you > hadn't, make sure you're doing an fxotu

Re: [asterisk-users] SIP registration

That doesn't seem to make any difference. I still get the "Not a local SIP domain" and I get this from the CLI: ast*CLI> sip show peers Name/username HostDyn Nat ACL Port Status 202(Unspecified)D 0Unmonitored 201

Re: [asterisk-users] Re: Polycom 601 loop

Nathan Bell wrote: > This is what I get from the asterisk CLI: > > ast*CLI> sip show peers > Name/username HostDyn Nat ACL Port Status > 202(Unspecified)D 0Unmonitored > 201(Unspecified)D

Re: [asterisk-users] Re: Polycom 601 loop

That seems to be the problem. In my dhcp settings I wasn't giving it the correct domain-name-servers option. I changed that and I changed the phones to use [EMAIL PROTECTED] instead of [EMAIL PROTECTED] and that seems to have taken care of it. Thanks for the help. Nathan Bell IT Engineer Du J

Re: [asterisk-users] SIP registration

The problem was on the polycom provisioning setup. In my dhcp settings I wasn't giving it the correct domain-name-servers option. I changed that and I changed the phones to use [EMAIL PROTECTED] instead of [EMAIL PROTECTED] and that seems to have taken care of it. Thanks for the help. Nathan

[asterisk-users] SIP REFER

Hello, I already wrote a message but maybe the problem I described was pretty unclear, so here a new Try. The thing is it is really urgent and I am not capable to resolve it for myself. Actually I am not a Asterisk Specialist, but I have to use an Asterisk Server for my thesis... So here the pro

Re: [asterisk-users] Doorphone

On 26 mar 2007, at 22.17, Ray Wadkins wrote: We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem

Re: [asterisk-users] Chan_cellphone and CentOS 4.x

Thanks for the tip, I got the device to pair, now it just won't stay connected. In Asterisk the CLI shows -- Bluetooth Device bruce has connected. -- Bluetooth Device bruce initialised and ready. -- Bluetooth Device bruce has disconnected, reason (104). Device bruce is a Motorola V3, wh

[asterisk-users] Server Recomendation

I am looking to install a system with 200 phones (polycom). There will be about 30-40 simultaneous calls. I am looking at the Dell 1950 with Quad 2.66, 2Gig RAM, Two 160 Gig SATA Drives (Mirrored with a Perc5 card), Dual Gig NIC, and RHEL 4.0. I will use two "gateways" for my PRI's and FXS Card

[asterisk-users] rx_fax and Asterisk 1.4.2

"0?s|40:2") in new stack -- Goto (incoming,fax,2) -- Executing [EMAIL PROTECTED]:2] Set("Zap/5-1", "TIMESTAMP="20070326-150200"") in new stack -- Executing [EMAIL PROTECTED]:3] Set("Zap/5-1", "FAXDIR=/var/spool/faxes&quo

Re: [asterisk-users] Re: Polycom 601 loop

Nathan Bell wrote: This is what I get from the asterisk CLI: ast*CLI> sip show peers Name/username HostDyn Nat ACL Port Status 202(Unspecified)D 0Unmonitored 201(Unspecified)D 0

Re: [asterisk-users] Server Recomendation

We have used the 1950's and 2950's in previous installations, and they work well. The main Dell issue is compatibility with the TE110P. PaulH On Mon, 2007-03-26 at 22:35 -0400, Forrest Beck wrote: > I am looking to install a system with 200 phones (polycom). There > will be about 30-40 simulta

Re: [asterisk-users] Doorphone

Ray Wadkins wrote: We have a doorphone device that's connected to our PBX. Currently, there's a special meetme conference that the phone connects to when the visitor presses zero. Users in the office can dial the meetme conference and get connected. The problem is that we can't send DTMF si

[asterisk-users] Refresher course needed!

Hello everyone My name is Brad, I am an old Asterisk Vet of the very early days just coming back to join the group. Ok, for starters, I feel like the "monkey with the light bulb" looking at extensions.conf and sip.conf. It has been some time. A friend ask me to set up a asterisk server that rec

Re: [asterisk-users] How is this feature called ?

Call Forward Override I guess. On 3/26/07, Olivier <[EMAIL PROTECTED]> wrote: Hi, Your colleague has forwarded his incoming calls to his secretary. How do you call the feature allowing you to circumvent your colleague call forward to make your colleague's phone ringing ? Best regards __

Re: [asterisk-users] SRTP vs ZRTP in Asterisk

My understanding based on the notes from the IETF meeting last week is that the SRTP Key Exchange group met and is considering two options. They are exploring DTLS and ZRTP. (There was a third option on the table which they are no longer exploring) The biggest issue with ZRTP seems to be non-tec

[asterisk-users] SIP Video Camera

Hi, Does anyone know of a Video Camera that is based on SIP? There are lots of Video Phones out there, but I can't seem to find a Video Camera. Thanks in advance. -kokmeng. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users m

[asterisk-users] how to define a pilot number

Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another