similar to: Asterisk/GXW410x IP Analog Gateway

Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 roy at

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

Dear All, We have a customer who is opening a new office in Dubai and we know that VoIP is blocked over there. Has anyone a solution to getting VoIP back out (we want interoffice calls back to the UK)? We we're thinking of IAX trunking, but not sure if that is blocked or just SIP etc. A VPN works, but is not great. We have seen: http://www.speed-voip.com/voiceguard.html At the moment it

Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm

Dear all, Just wanted to know if any one had deployed the Grandstream GXW 4024 yet. Wanted to hear any feedback and/or problems with this unit that you may have experienced. Thank you. -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd www.emergen.biz

I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey

somebody knows some application web that allows me to call to my internal extensions of my asterisk, example click to call. I was proving the click to call of this example but it doesn't work http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html greeting rickygm

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

Hi, I had asterisk 1.4.17 with the extensions which is configured in the sip.conf it was working fine. Now I am having the requirement to authenticate the SIP users through the OpenLDAP not through the sip.conf. Steps I have done : Did a check out by using the following command, http://svn.digium.com/svn/asterisk/trunk. [^] then given configure, make , make install. and taken the sample ldap

Hello Has someone written a web page (preferably PHP) that simply shows what extensions are currently online? Thank you.

Hello, How can I make a live chat (mainly text, but with voice/video chat if possible) interacting with asterisk? Can asterisk control simultaneously the queue between people calling by phone and people by web chat? At my work, there is a call center using asterisk to control the queue of the clients (by phone) already. This part is ok. But now I need to make a chat room at the website

Greetings list, I've been asked to provide a system for 200 extensions, most of which will be existing analogue POTS handsets, not IP handsets. I've not really had any experience with large channel banks in the past (since most of our deployments are strictly IP-only to the desk), so I'm at a loss as to which ones are worth looking at. If anyone's had experience using channel

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Has anyone here implemented "Ring back when free" in Asterisk? The way it works in the UK is as follows: 1. A calls B. B is engaged (busy). 2. A hears "The number you called is busy. To use ringback, press 5" 3. A presses 5, and hears "Your ringback request has been accepted". 4. A hangs up. 5. Later, B hangs up. The system then calls A (if A is now busy, it

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When

Hello, I got Quintum A800 with the P5-2-1 firmware. I configure my asterisk trunk as followed: [SIP_BD1] type=peer qualify=yes host=192.168.0.254 disallow=all context=from-pstn allow=h723 and inside the quantum I change the config sip useragent to 5060. Up to this part if I run sip show peers, I got: asterisk1*CLI> sip show peers Name/username????????????? Host??????????? Dyn Nat ACL

As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone