search for: yurchenko

Displaying 18 results from an estimated 18 matches for "yurchenko".

2003 Jun 27
2
Making calls from snom 100
...------------------------------------------------------- the "sip debug" dump: ------------------------------------------------------------------------------ INVITE sip:172.20.0.170 SIP/2.0 Via: SIP/2.0/UDP 172.22.0.199:5060;branch=z9hG4bK-ubuamvlt6h17 Max-Forwards: 70 From: "Anton Yurchenko" <sip:100@172.20.0.170>;tag=i7n7jzxqp3 To: <sip:200@172.20.0.170;user=phone> Call-ID: 3c267e34226a-wohzkq5t9qqd@172.22.0.199 CSeq: 1 INVITE Route: <sip:200@172.20.0.170;user=phone> Contact: <sip:100@172.22.0.199:5060> User-Agent: snom Version 1.15u Accept-Language: en A...
2003 Nov 28
4
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<phila@dg.net.ua> Digital Generation
2003 Nov 24
3
strange SIP authentication/authorization behaviour
...roup=1 and this user has a wrong password then calls are denied, but when I just change the userID on the phone to a nonexistant for example 110, the calls go through ! though I see on the console messages about wrong authentication. I`m running a CVS version from friday. Thanks, -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Dec 12
2
Dlink DG-104SH
Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Dec 02
2
incominglimit stuck in app_queue
...n the console resets this and they are again available for working. anybody see this.? i`m running todays CVS, and phones are Grandstream BT-101, with firmware : Program--1.0.3.81 Bootloader--1.0.0.7 HTML--1.0.0.18 email me if you need anymore details. thanks a lot in advance -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Dec 23
2
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<phila@dg.net.ua> Digital Generation
2003 Nov 28
4
Mute button in Grandstream?
Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Jul 03
3
Using switch =>
...20 ; How frequently to send trunk msgs (in ms) ; tos=lowdelay register => phila:[test]@172.20.0.170 ; ; [hurricane] type=friend host=dynamic trunk=yes ; Use IAX2 trunking with this host context=default auth=rsa inkeys=hurricane outkeys=test ; ------------------------------- -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Dec 25
1
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<phila@dg.net.ua> Digital Generation
2003 Jul 28
2
"immediate=yes or Compleate recieved" with intcoming calls with new CVS
...ungup 'Zap/1-1' < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 140/0x8C) (Originator) < Message type: RELEASE COMPLETE (90) > Ext: 1 Cause: Unallocated (unassigned) num exten => 2382031,1,Dial(SIP/100,20,t) -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Jun 23
1
Setting up the E100P
...ss its euroisdn ? ) anybody could guess what is the problem? The admin that runs the Cisco says that signalyng should be PRI , and there is an option for pri signaling in zapata.conf, but the zaptel,conf doesnt have it and so I`d get up with a mismatch, and zasterisk would not start. -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Dec 29
1
transfer with MGCP
...dpoint 'aaln/4@dg104s-0' observed 'hu' -- MGCP handle_request(aaln/4@dg104s-0) ast_channel already destroyed -- Endpoint 'aaln/3@dg104s-1' observed 'hu' == Spawn extension (icg, 154, 2) exited non-zero on 'MGCP/aaln/3@dg104s-1' Thanks. -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2003 Jun 25
6
snom 100 and GSM codec
...console: WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames -- Anton Yurchenko<phila@dg.net.ua> Digital Generation
2002 Apr 26
9
[Fwd: Re: borrowing only from parent]
Martin Devera wrote: > If you read the manual, the algorithm will not work correctly > with {,c}burst < MTU ... > devik > I just tried to change {,c}burst to 1600, or leaving them by default but no visible result. here is the latest tc -s -d class show dev eth0 class htb 1:101 parent 1:1 prio 0 rate 40Kbit ceil 40Kbit burst 1599b/8 mpu 0b cburst 1599b/8 mpu 0b quantum 512 level
2003 Nov 27
1
App queue and all Agent busy
...when both Agents are busy then still the called party does not get a busy signal. What I`d want is when both Agents are 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<phila@dg.net.ua> Digital Generation
2003 Dec 23
2
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<phila@dg.net.ua> Digital Generation
2002 May 16
1
Install Question
...s on this Would appreciate it if someone could give me a few pointers, or point me to a doc...there''s very little in the LARC Howto. Thanks in advance Justin Owens -----Original Message----- From: Herman Cremer [mailto:admin@help.co.za] Sent: 15 May 2002 07:58 To: Martin Devera; Anton Yurchenko Cc: lartc@mailman.ds9a.nl Subject: RE: [Fwd: Re: [LARTC] borrowing only from parent] Question : > > Has anyone ever managed to compile iproute2 from source, > > running a 2.4.x kernel ? I have also managed to solve it in another way.. Here is how ... Solution 1: Well I have man...
2004 Jan 26
0
Anyone run * on OS X ?
...ject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Has Nufone gone belly-up (Steve Underwood) 2. SIP - fax / voicemail (Dawid Mielnik) 3. Re: Has Nufone gone belly-up (Girish Gopinath) 4. app_queue and dialplan (Anton Yurchenko) 5. Know if a call is answered (Asterisk List) 6. Re: rc.local dont works (Jeroen) 7. Re[2]: [Asterisk-Users] Sipura 2000 Transmit Issues? No Sound being passed to caller (Frankie Gravato) 8. Re: OH323 doesnt hear ringing (Michael Manousos) 9. RE: Asterisk Indications (Christopher Le...