similar to: Looking for a way to show caller id information on the desktop

Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090708/cccd4587/attachment.htm

We have users with Cisco 7900 phones running sip. When user A calls user B, we want user B's name to appear on user A's phone. It shows the extension they call, but not the internal name of the called user. Is this possible? We have some people that used to be on an MGCP based system and they would get the callee's name popup on their phone when they called someone. I

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

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

I have setup a few Asterisk systems for customers using Digium TDM400 cards and Aastra phones. No problems with sound quality at all except at this one site. Every time I try their system I don't hear any problems but they tell me that it is really bad. They describe it a a loud scratching sound. Are there any tests that can be done to pinpoint the problem? Has anyone seen this before? Are

Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging

Hello all, I need to install asterisk for 900 sip users with 2 PRI ports. It is posible to handle this number of calls/extensions with only one asterisk machine? Which is the best way to install that? two asterisk with openser. One asterisk with openser ..... Is it necesary run a SER server on this enviroment? Any clue will be welcomed. Thanks in advance. VoipCrazy

Hi everyone. We have an asterisk server in our main office and phones at each remote site. The remote offices are connected via a MPLS which, to my knowledge has no natting going on. The problem I have is that any call from a remote phone to a remote phone (even on the same remote lan) results in no audio. If I make a call from the same LAN the asterisk server is on, to one of these remote

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }

Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers

Hi guys, Can I use a Sangoma a101 to interface a legacy pbx to an Asterisk server? The pbx doesn't have sip and I want to come in off of a sip trunk and interface with the older system. I know I can use a pri card to hook in to the phone network, but can I use this same card to feed back the signaling as if I were the phone company to the older system? Thanks, Tom

Hi, I need to connect 2 asterisks in 2 different countries (A and B) for one company so it's possible to make connections between the 2 offices. For connectivity reasons (NAT traversal) i want to connect the 2 asterisk with IAX so that when a user on office A connects via SIP to user on office B the call is going trought IAX channel. Can anyone give me an ideia how to accomplish

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and

Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like "555-555-1234 at 192.168.1.10" where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

Now that we have voicemail working, people have asked to be able to dial in externally and be able to access their voicemail. My dial plan is simple, after ringing a few extensions for some time, it goes to voicemail. What needs to happen to allow for someone to switch out of this into Voicemailmain in such a fashion that an external inbound caller wouldn't at least hear the option? Can the

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a

Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell