similar to: Distinctive ring

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

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

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",

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'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

Cisco 7960 has six buttons/lines. Can some of them be configured for fast dialing? If it can't be configured on the phone, how can I configure it on Asterisk? -- 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

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

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

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! 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 use asterisk 1.2.5 and h323 that comes with addons 1.2.1. Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and tries to make attendant transfer to person B (local SIP phone). They speak. Then A hangs up. Call form h323 trunk doesn't get to person B. This is what I get on CLI. -- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663 --

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 tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi group! How to set language for queue? I have several queue's. In every queue, agents speaks different language. I need to announce queue-youarenext and similar on different languages. This is what I have in my extensions.conf and it does set language, but when calls enters queue, it doesn't use that language. exten => 313,1,Answer exten => 313,n,Set(LANGUAGE()=de) exten =>

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 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

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