similar to: Making calls from snom 100

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

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

When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 <109> callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

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

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

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

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

Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn signalling=fxs_ks usecallerid=yes hidecallerid=no

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 and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

Anybody has figured out why asterisk + snom have such bad quality using GSM? When I use GSM I see such messages dumped on asterisk 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

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

I have a queue defined as [blabla] member = SIP/101 member = SIP/102 and in extensions.conf this: exten => 101,1,Queue(blabla,t) exten => 101,102,Congestion but 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 (

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

Sorry for asking such basic questions but I haven''t used iproute or htb before. I am using RedHat 7.1 (2.4 kernel) and I want to use HTB3.3 and iproute2-2.4.7 Before I make and install iproute do I need to do something with the HTB files...I can''t find any docs on this Would appreciate it if someone could give me a few pointers, or point me to a doc...there''s very

With the 1U Apple G5 server at a good reliability/storage/pricepoint level... got to thinking about compiling Asterisk on OS X.. at least for SIP phone call switching, voicemail, etc. Has anybody attempted this? Email me off list if this is too dev-heavy for the user list. Thanks, Ted W -----Original Message----- From: asterisk-users-request@lists.digium.com

Hi all, I've just been setting up such a phone with * and I'm facing some problems. At this time there is no customized extensions & others, so I use extension 8500 (voicemail) in default config. What I get when this extension is called from the SNOM is pasted below. The phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in the "Contact"