similar to: Asterisk as outbound proxy

Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

Hi, I have a dlink dvg-1120M (mgcp) box that i will like to use with asterisk. Is it possible? has anyone done that? Here's a link to the product page at dlink. http://support.dlink.com/products/view.asp?productid=DVG%2D1120M Also, does anyone has or know where to get the firmware for Dlink DVG-1120S (sip model)? thanks. -- Zahid

OpenSSH Unix Dev, Mac OS X 10.5 recently shipped with OpenSSH 4.5p1. This build includes a number of patches, some general bug fixes and some platform- specific fixes and enhancements. These patches are available from our open source site (http://www.opensource.apple.com/darwinsource/10.5/OpenSSH-87/ ). Following is a brief description of each patch. We'd be more than happy to

I upgraded a DVG-1120M to a DVG-1120S. Everything works great, but I'm having some caller ID issues on incoming calls sent to the SIP device. Using debug on the device, the caller ID looks fine - just as I set it in Asterisk. However, the phone is showing "CID TRANSMISSION ERROR". Should I check the RX and TX gain levels? Try another phone? Any ideas would be much appreciated.

I had just found a Dlink DVG-1120 on ebay and I'm curious if anyone has used you it with asterisk. They were only $65. I have tested with the Linksys Pap2 and found that box to be fairly nice except for a lot of backgound/white noise. I was wondering if any else had experienced that? Let me know if I've wasted $65 on the Dlink and also if you had similar experience with white noise on

I friend gave me his DVG-1120s after he realized that AT&Ts callVantage stuff would not work for him. It appears to be running a SIP version of firmware, however, it downloads an XML configuration file via SSL from AT&T. I cannot find a way to manually configure the VOIP portion of the unit via the GUI. I contacted D-Link to get an example configuration file so I could get it working

Primus in Canada has launched a SIP-based service to replace your business and residential POTS lines with a VoIP version. It's called TalkBroadband and it looks killer: http://www.primus.ca/en/residential/talkbroadband/index.html Basic service for $20 Cdn a month!! Local number portability!! Cheapo Primus LD rates!! They don't care where geographically you plug it in!! When you sign

I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP. David >>> asterisk-users@eol.ca 1/21/2004 6:39:34 AM >>> I'm not sure Primus

Hello, What is the set of configuration that should be done in the Asterisk 1.0.8 using FreePBX that can allow a simple video call between two extensions? Thanks in advance. [http://www.ericsson.com/shared/images/Email_line.gif] JULIO ARAUJO TE ENGINEER MS Ericsson ITTE & Test Environment S?o Jose dos Campos, Brazil Phone +551239084121 SMS/MMS +551281150089 julio.araujo at

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

If you look at the specs on the Dlink box that Primus gives you, you will see that it is SIP. I am sure Primus has a SIP platform because we have played with it. We managed to use it on MSN's SIP phone as well as couple Zultys ZIP2x2 hard phones. Their PC-Phone app is also a SIP soft phone. If you are registering to sip.iprimus.net then it is definitely their SIP platyform not MGCP.

Hello fellows!!! I'm having problems with Ericsson Dialog 3221 phone and DISA. I've configured an extension to test DISA and it work properly with all other phones, but freeze with the mentioned phone. Here is my extension: exten => 105,1,Answer exten => 105,2,Set(TIMEOUT(digit)tting =5) exten => 105,3,Set(TIMEOUT(response)=10) exten =>

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

Hello, Our asterisk is connected to an ericsson pbx by PRI. What i want is the asterisk clients should call operator numbers by dialing 0 But, when a call is made to ericsson via number 0, it assumes that the call is made from outside, so it doesnt allow to be dialed. There are 3 real operator extensions which is grouped by ericsson for operators. Lets assume 1111 1112 1113. What i want to know

Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the

Hello, from time to time I keep getting problems with some emails causing signal 6. I've already reported those, but it seems not to be easy to find the cause. From the logs, it seems to occur in sieve implementation. I've checked the email envelopes tody by accident, probably this part of my telnet session might help: a11 fetch 1 all * 1 FETCH (RFC822.SIZE 16750 INTERNALDATE

Hi all, I was using dahdi 1.6.2.0.9 version for a long time. We decided to upgrade to 1.6.2.22 a few days ago. After that we started to have some problems with dahdi channels. PS:DAHDI Version: 2.6.0 Echo Canceller: HWEC, MG2 We have 2 PRIs between Ericsson pbx and asterisk and a sip trunk for outside calls. At begining everything works fine but in a few hours, calls from asterisk to ericsson

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Editing subject line to reflect current status. On 1/26/06, Nilesh Londhe <lvnilesh@gmail.com> wrote: > Since T616 is not answering (and incoming calls are going to Cingular > voicemail after 30 sec,) I suspect the problem focus area is... > > > -- Executing Answer("BLT/T616", "") in new stack > > Is

We have a Dlink DVG-1120M and were surprised that it was able to handle 2 simultaneous conversations to 2 seperate phones using only 1 MAC address and 1 IP address. So we asked ourselves..why can't other 1 MAC/1IP devices do this as well? I have a Grandstream 486 that has 1IP and 1MAC. But I don't see anywhere in sip.conf to add a second line to a device. Is this possible? Can this only