similar to: Huawei EP201S

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

Hey, does anyone know why i'd be receiving: Aug 28 19:40:04 DEBUG[1875]: ##### Testing 66.27.233.241 with 10.0.10.0 Aug 28 19:40:04 DEBUG[1875]: Target address 66.27.233.241 is not local, substituting externip I get tons of them, usually when the phone is registering/calling/receiving calls. Thanks! Chris

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 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 have a client that wants a function that will automatically logout an agent from a queue if they do not answer a call. This would prevent future calls from being sent to that phone if the agent forgot to logout. Any ideas? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>

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 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 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

When I start * I get 6 mpg123 processes start as well. Is this normal? Often after a couple of days these mpg123 processes start to consume cpu and I have to kill them off. I do not have a sound card in the server and I have noload => chan_oss.so Simon

Hi Folks, I would like to recover all information about the calls, incoming calls, call time, call history, etc in a Web Format, are there some open source aplication for Asterisk that be easier for use. Pls anything suggestion will be very appreciate. Thanks Rgds. -- Omar E.P.T ----------------- Certified Networking Professionals make better Connections! http://omarept.blogspot.com/

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

Hello all, Just to update, Instruction's can be found at www.omniis.com/asttapi, including where to download it from. This is update 0.02, this now includes a little feedback from Asterisk so that when click to dial has occurred then it is indicated at the start and the end of the call. Now working on inbound calls. Any question, please send to me. Regards Nick

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

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Greetings, I am cutting my teeth with SIP phones and my first issue is getting a Cisco 7940 to Authenticate with my VoIP provider (BBTelsys). I did read some notes on the vo-ip website about 7.5 being the better firmware version. Has anyone had trouble with 8.2 and SIP registering? Should I just downgrade to 7.5 and give it a go? I think SIP uses UDP 5060 correct? The phone is behind a

Hello, We have several regional asterisk's connected to a central one making the the PRI calls through a TE410P card. When using SetCallingPres(prohibited) on a call at the regional level, that setting it not forwarded to the central asterisk and the call is made as if no callerid had been sent: the telco substitutes the network number. Using SetCallingPres(prohibited) on the central

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.0/639 - Release Date: 18/01/2007 18.47 -------------- next part

I am looking for an ATA that has had very reliable results when passing FAX over IP. I was thinking of testing the Cisco (not Linksys) ATA 186 I1, ATA 186 I2, ATA 188 I1. This is what I'm looking for: FAX -> PTSN -> through Asterisk -> ATA -> Fax Machine. I have QoS from PSTN entry to ATA on the network so I can assure precedence. What has everyone out there been using