similar to: Cisco 7960 - Fast dial

Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have

This is one long letter about T.38 and Asterisk. I hope it will help me, and lots of other Asterisk users to understand some T.38 problems with Asterisk. This is my situation: I have Panasonic DX600 FAX machine. It's connected to Asterisk 1.2.13 thru ATA adapter (I have used both, Cisco 186 and Grandstream HandyTone 386). Asterisk is connected with my SIP provider. That link that my provider

I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [root@tomo ~]# ping sip1.voipbuster.com PING sip1.voipbuster.com

On one installation (* 1.2.13) Pickup doesn't work. This is what I have in extensions.conf exten => _**2X,1,Pickup(${EXTEN:2}&8${EXTEN:3}&t&uevents) exten => _**2X,n,Hangup This is what I get on CLI -- Executing NoOp("mISDN/3-1", "incoming-beronet 80 - dolazni poziv s broja 270248") in new stack -- Executing LookupCIDName("mISDN/3-1",

Is uniqueid globally unique? I have three Asterisk installations and I need to store data from all of them in same database, in same table. Will this uniqueid field be unique? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr -------------- next part -------------- A

I'm having trouble installing Beronet BN4S0 card. I have downloaded instructions from here http://www.beronet.com/download/card_installation_guide.pdf And when I download install-misdn-mqueue[1].tar.gz I untar it and execute "make" and "make install". This is the output that I get. [root@nbirma1 install-misdn-mqueue]# make install make -C app_bundle make[1]: Entering

Does anybody use Intel S3000AHLX board with Digium TE110P E1 card? Have you experienced any problems? I'm planning following configuration, so I would appreciate any experience both positive and negative. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

Hi list! I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5 (I know, I should upgrade) and in dial plan I have: exten => _64X,n,Set(_ALERT_INFO=Chirp2) exten => _64X,n,Dial(SIP/${EXTEN},30,wWtT) On Cisco in Settings => Ring type I have "Chirp1" and "Chirp2". By default phone is ringing sound "Chirp1". For internal calls

Has anybody used Huawei EP201S IP phone? I need 9 SIP phones that cost around 100USD, and those phones are one of options. Can anybody suggest anything else that costs around 100USD? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 SIP: tomo@pbx.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call

I have asterisk connected to E1 interface with Digium TE110P. I have Cisco ATA 186 and fax pass thru well. I have tried to establish modem connection (from computer connected to ATA => SIP => * => E1 => Telco => pstn => another modem) and I do connect (at 14,400) but connection end after a minute. How to establish successfully modem connection? Has anybody tried innovaphone IP21

Hi list! I'm trying to setup Digium TE205P E1 card to work with my provider (and Ericsson BP 250 - this comes later). When I execute "pri show span 1" this is what I get. pbxzg1*CLI> zap show status Description Alarms IRQ bpviol CRC4 T2XXP (PCI) Card 0 Span 1 OK 0 0 0 T2XXP (PCI) Card 0 Span 2 UNCONFIGUR 0 0 0 pbxzg1*CLI> pri show span 1 Primary D-channel: 16 Status:

I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of

One simple question. I'm using asterisk 1.2.1, can one agent be defined in more than one group? Example: group=1 ; queue1 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 403,403,Sasa Juginovic group=2 ; queue2 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 404,404,Marija Bilic agent => 405,405,Ana

I have defined 4 queue's. Is there any way to check is there any agent logged in any of those queue's? What I would like to do is to check if there is any agent in any of queue's and if there is, then I'll will transfer a call to that queue, it there isn't I would like to do something else with a call. Thank you for your time. -- Tomislav Par?ina Lama Computers Split

Hi people, pls does anybody know what "(T)" and "(D)" letter means? server3*CLI> iax2 show peers Name/Username Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

I am seeing the following in my log file (standard trixbox install). One seems to be complaining about an error in the dialplan but it won't tell me what file or what line. The other (maybe related) is complaining about a channel lock. How to do go about trying to figure out what the problem is and how to solve it? ---------------Logfile-------------------------------------------- Nov 14

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded