similar to: strange ring

Hello that might would be an easy question for someone, but im in doubt Is there any possibility to pass a call from one asterisk to another and then to ZAP channel. For instance I have "A" asterisk with numbering 45670 "B" asterisk with numbering 45680 second asterisk has TE110P card with single PRI port connected to Siemens EWSD. When I originate call from asterisk

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

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

So, I recently acquired an old DVG-1120M that was previously used with AT&T CallVantage service. It currently uses MGCP. I see online that there is a SIP version out there (DVG-1120S), which is essentially a DVG-1120M reflashed with the SIP firmware and the 'M' scraped off in favor of an 'S'. I've also seen people on this list saying that they flashed their M with S

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

Hi Everyone I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box that needs to be secured at all times. Currently it's not connected to the internet. If it were connected, I would have iptables block any and all traffic from outside but I want a single device - Linksys PAP2T - to be able to connect back to the server. That is a stand alone device and doesn't support

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.

Thu, 06 May 2004 12:31:33 -0500 there was an e-mail to the list from Isaac McDonald suggesting to re-flash the DVG-1120M gateway with the DVG-1120S code to have the gateway login to *. I have been looking and can seem to find the flash code. D-Link tech support doesn't know much about the box. Any sources for the TFTP file would be appreciated. Once this is running SIP will I need the xml

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

Hello !!! I'm trying to setting up my DVG-2032S Voip Gateway to use Asterisk as outbound proxy, that's because I already have this gateway before to begin to play with Asterisk. Every time when I enable the OutBound Proxy option and call from my Ericsson PBX I got the follow message in DVG-2032S System Information: Hook off, and nothing in Asterisk log or in the console. I've

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

Hello, Has anyone experienced a segmentation fault in asterisk for using G729 against an AS5300 in SIP ? I'm having this problem. Also, any other codec I use gives me incompatible media (can be read in SIP DEBUG messages). AS5300 configured for multiple codecs, so is Asterisk. Tried G711u/A G723 and G.729. Any clues ? Regards, Jorge A. Info: Asterisk ver 1.0.7 stable Using AMPortal

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

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

Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf