[Asterisk-Users] Pingtel registration failing

Hello, I have a Pingtel Xpressa and trying to get it working with *. When the phone tries to register, it sends out a REGISTER request and * replies with PROXY AUTHENTICATION but phone never replies back with the right info and just sends REGISTER again and again. This is what Pingtel support

[Asterisk-Users] permit/deny in sip.conf not working

kinda defeats the purpose. Is this a bug or am I missing something? The phone is on 192.168.0.0/24 so it should deny it with the config snip I have below. I`m running: CVS-04/22/04-09:13:06 here is a clip from the config: [11] context=local callerid=Anton Yurchenko 11 type=friend username=11 secret

[Asterisk-Users] permit/deny in sip.conf not working

kinda defeats the purpose. Is this a bug or am I missing something? The phone is on 192.168.0.0/24 so it should deny it with the config snip I have below. I`m running: CVS-04/22/04-09:13:06 here is a clip from the config: [11] context=local callerid=Anton Yurchenko 11 type=friend username=11 secret

[Asterisk-Users] long delay before asterisk returns 486 busy with sip

for snom and grandstream phones. and this is verivied with ethereal. I`m running CVS-12/01/03-14:50:57 version. Anybody else see this happening? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] ATA in MGCP sometimes dropping calls

to transfer to but it is busy, and when they return to the first call it is dropped. I have not been able to reproduce this and get some mgcp debug messages. I`ve tried the 2.15 and 3.0 firmware versions. Anybody else experincing this? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital

[Asterisk-Users] asterisk as B2BUA with SER?

Hello, Has anybody done this? Do you have any pointers on this or maybe even a howto. Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

[Asterisk-Users] app_queue and dialplan

the announce message and and no music on hold is played and again the announce message is played. somehow the music on lod doesn start. What am I doing wrong? I run version CVS-12/01/03-14:50:57 Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

[Asterisk-Users] app_queue and dialplan

the announce message and and no music on hold is played and again the announce message is played. somehow the music on lod doesn start. What am I doing wrong? I run version CVS-12/01/03-14:50:57 Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

[Asterisk-Users] Description of Manager events

Hello, does anybody have a list of asterisk Manager events and what they mean? For examples such events as Rename? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Dialing chan_local. Was: [Asterisk-Users] return of the transfer to a busy number

am I doing wrong -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] call pickup via *8 from ata186 (SIP)

-- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)

CW_ASN wrote: It works for me with sip 2.15, 2.16.x and 3 versions. I have FW version : *ata0009e88e33cd* Version: v2.15 ata18x (Build 020927a) MAC: 0.9.232.142.51.205 what other parameters should I use? What AudioMode? thanks - Original Message - From: Anton Yurchenko [EMAIL

[Asterisk-Users] transfer with MGCP

. -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] transfer with MGCP

Pavel Litvinenko wrote: Anton Yurchenko wrote: Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long

Re: [Asterisk-Users] transfer with MGCP

Pavel Litvinenko wrote: Anton Yurchenko wrote: Pavel Litvinenko wrote: Anton Yurchenko wrote: Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash

Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)

Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, December 29, 2003 8:30 AM Subject: Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP) CW_ASN wrote: It works for me with sip 2.15, 2.16.x and 3 versions. I have FW version

Re: [Asterisk-Users] return of the transfer to a busy number

Eric Wieling wrote: Anton Yurchenko wrote: Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. exten = 123,1,Dial(Zap/5,30) ; ring Zap/5 for 30 seconds exten = 123,2,VoiceMailMain(u123) ; No answer

Re: [Asterisk-Users] return of the transfer to a busy number

Anton Yurchenko wrote: Eric Wieling wrote: Anton Yurchenko wrote: Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. exten = 123,1,Dial(Zap/5,30) ; ring Zap/5 for 30 seconds exten = 123,2

[Asterisk-Users] return of the transfer to a busy number

Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] call pickup via *8 from ata186 (SIP)

Hello, Does call pickup works with subj? at the same pbx it works with MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. Anyone have it working? Also it seems that when typing reload on the console, the asterisk doesnt reread the mgcp.conf. Thanks -- Anton Yurchenko[EMAIL

[Asterisk-Users] Asterisk + CRM

Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

[Asterisk-Users] gnophone transfer

hello, Is there a way to transfer the call via gnophone, without calling other user and pressing conf on both calls, it seems that all traffic is still going through the gnophone, not that optimal i guess. thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

[Asterisk-Users] MGCP call waiting disable?

busy, and to disable call waiting? thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Seting callerID on outgoing calls

callerid=888 channel=11-20 When I call to outside the caller id shows only the same one Thanks, -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] How to return a transfered call

these both the same ; callgroup=1 pickupgroup=1 So if you transferred the call to can extension and you were in the same group you could dial *8 and get the call back. Sorry I didnt mention that, what if the line I`m transfering too is busy? How do I return the call then? --- Anton Yurchenko [EMAIL

[Asterisk-Users] Dlink DG-104SH

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] incominglimit stuck in app_queue

registrations and timeouts. Please test and provide some feedback. well its been whole 24hours and no problem has been observed, we`ll see, if you dont hear about this, then consider this fixed. Thank you very much for your work on app_queue, those fixes came just in time ;) -- Anton Yurchenko[EMAIL

[Asterisk-Users] app_queue different behaviour

. And if the caller calls , and there are no free operators , some message like please wait for next avalable operator and them the music on hold start. thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] Transfer via # on Grandstream not always working

see this happening also? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] app_queue different behaviour

or not. Thank you Good luck! Michiel Anton Yurchenko wrote: Hello, is there a way to make app queue to first try to ring the agents and start music on hold only when they are all talking to other callers? So when the caller calls, and there are free operators he hears ringing, and * is not picking

[Asterisk-Users] app_queue and CDR

this exten = 101, 1, Answer exten = 101, 2, Queue( phila) the line is answered and musiconhold plays. thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] incominglimit stuck in app_queue

in advance -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] incominglimit stuck in app_queue

in advance -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] app_queue behavior followup

Yurchenko Sent: Sunday, November 30, 2003 8:33 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] app_queue behavior followup Anton Yurchenko wrote: also if I build my dialplan like : exten = 101,1,Answer exten = 101,2,Queue(phila) The musionhold plays only until the track is finished

Re: [Asterisk-Users] app_queue behavior followup

Anton Yurchenko wrote: Joe Dennick wrote: I think you need to better define your Queue Environment in extensions.conf. Below is what I've got in mine, and it seems to work quite well: exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4

Re: [Asterisk-Users] call waiting disable in sip

Paul Liew wrote: - Original Message - From: Anton Yurchenko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 29, 2003 3:34 AM Subject: Re: [Asterisk-Users] call waiting disable in sip what would happend if all operators are busy? would app_queue exit? would

Re: [Asterisk-Users] call waiting disable in sip

Walker Haddock wrote: On Mon, Dec 01, 2003 at 03:33:50PM +0200, Anton Yurchenko wrote: I have a problem, when caller is in Queue and the operator is busy answering other call he/she still hears the call waiting signal. I have the latest cvs and incominglimit is set to 1. But here is what

[Asterisk-Users] app_queue behavior

callerid=ipphone1 200 callgroup=1 pickupgroup=1 incominglimit=1 thank in advance -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] app_queue behavior followup

Anton Yurchenko wrote: also if I build my dialplan like : exten = 101,1,Answer exten = 101,2,Queue(phila) The musionhold plays only until the track is finished, and then it hangsup. How to make it loop? -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

[Asterisk-Users] call waiting disable in sip

Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks -- Anton Yurchenko[EMAIL

Re: [Asterisk-Users] call waiting disable in sip

Patrick wrote: On Fri, 2003-11-28 at 09:14, Anton Yurchenko wrote: Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find

[Asterisk-Users] App queue and all Agent busy

that the caller gets a busy, not the long tones, like the phone is ringing, but nobody answers it ( this is how this works now) Any Ideas? Thanks in advance -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED

Re: [Asterisk-Users] App queue and all Agent busy

functionality, I can setup a prompt, like all operators are busy and such. thanks Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED

Re: [Asterisk-Users] App queue and all Agent busy

Anton Yurchenko wrote: I just wanted to say, that I patched the code like I wrote below, and it works. When all the operators are busy, then it drops to priority + 101. If that would break something please write me ASAP ;) Philipp von Klitzing wrote: Hi! but when both Agents are busy

[Asterisk-Users] strange SIP authentication/authorization behaviour

. Thanks, -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] snom100(with latest firmware) screeching noise when doing transfers,

month old CVS, and the todays version too Thnx -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Compleate recieved

Is there any way to find out why this happens? why do I get complete recived? see my previous post -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo

[Asterisk-Users] immediate=yes or Compleate recieved with intcoming calls with newCVS

: RELEASE COMPLETE (90) Ext: 1 Cause: Unallocated (unassigned) num exten = 2382031,1,Dial(SIP/100,20,t) -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http

Re: [Asterisk-Users] immediate=yes or Compleate recieved with intcomingcalls with new CVS

Anton Yurchenko wrote: the thing seems to be in the chan_zap.c doing a diff from the version that I have working I see this change: - @@ -5607,10 +5609,10 @@ strcpy(pri-pvt[chan]-callerid

Re: [Asterisk-Users] Using switch =

Brancaleoni Matteo wrote: a stupid question... have you started the key from the private keys by issuing init keys on the console ? yep I did that, still no luck Matteo. Il gio, 2003-07-03 alle 14:46, Anton Yurchenko ha scritto: Mark Spencer wrote: Looks as though there may

Re: [Asterisk-Users] Using switch = BUG?

Anton Yurchenko wrote: Mark Spencer wrote: Looks as though there may be a problem with RSA key authentication and switch for some reason... Does it behave different with either no authentication, or password authentication? Mark OK I seem to not have problems using switch = with other then rsa

[Asterisk-Users] Using switch =

; How frequently to send trunk msgs (in ms) ; tos=lowdelay register = phila:[EMAIL PROTECTED] ; ; [hurricane] type=friend host=dynamic trunk=yes ; Use IAX2 trunking with this host context=default auth=rsa inkeys=hurricane outkeys=test ; --- -- Anton

Re: [Asterisk-Users] Using switch =

as a follow up: when I make a call to the extension on the other box designated as switch, I see packets going to to iax port on the switch box but I dont see any relies from it. -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk

Re: [Asterisk-Users] Using switch =

? and while i`m at asking you questions is iaxtel.com ? thanks a lot for you work, * is the best -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman

Re: [Asterisk-Users] Enhanced queue app

://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Making calls from snom 100

From: Anton Yurchenko sip:[EMAIL PROTECTED];tag=i7n7jzxqp3 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Route: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: snom Version 1.15u Accept-Language: en Accept: application/sdp Allow

Re: [Asterisk-Users] Making calls from snom 100

Anton Yurchenko wrote: Hello, The Issue is fixed by setting in snom100 under Settings-SIP- Stack treat as: to address instead of route. Than happend becouse somebody has been plaing with the phones without me :) I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial

Re: [Asterisk-Users] snom 100 and GSM codec

are porting the software from the 200 to the 100/105 and are close to releasing something new. Please give as some more time. Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton

[Asterisk-Users] snom 100 and GSM codec

to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com

Re: [Asterisk-Users] Setting up the E100P

anybody know what do all this stuff in zttool mean? Thnks!! -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Calls from PSTN - SIP

proxy, becouse it`ll allow much richer usages ot the whole SIP features and allow for more scalability. But that cancels out registering/authorization on asterisk. -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list

Re: [Asterisk-Users] Experment: Turn this thread SNOM mini howto.

[EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation

Re: [Asterisk-Users] username option in sip.conf

but changing to username=phila doesnt help. only when its like [phila] ... .. it works. Username is not taken into account it seems. thnks roy On Monday 24 March 2003 13:58, Anton Yurchenko wrote: Hello, what purpose does the username option for a peer serves in sip.conf? I saw a mention the you

Re: [Asterisk-Users] username option in sip.conf

but changing to username=phila doesnt help. On console I see things like: NOTICE[122896]: File chan_sip.c, Line 3629 (handle_request): Registration from 'Anton Yurchenko sip:[EMAIL PROTECTED]' failed for '172.20.0.199' only when its like [phila] ... .. it works. Username is not taken into account

Re: [Asterisk-Users] username option in sip.conf

Anton Yurchenko wrote: Roy Sigurd Karlsbakk wrote: I beleive the username should be without the @xxx anyway I just read a draft of Asterisk Handbook2 and here what it sais about the username option: username: username Asterisk tries to connect when for some reason it differs from the one

[Asterisk-Users] about new sip.conf options

Hello, I see on the mailing all kinds of new sip.conf options appearing like dtmfmode , etc. Could anyone describe them, anything that is not in the version 2 Asterisk Handbook, the sip.conf.pdf. Thanks, -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation