similar to: California based PSTN connections

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Going over the needs of any PBX that replaces our current system (working toward Asterisk) and have VM functionality question. Currently when someone leaves a voice mail for a sales person (Who is in the field) the system takes the VM and then in turn dials over a POTS line and pages the sales person notifying them of a VM (Does not deliver the message-just notifies) Is this possible with

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you

Any Recommendations on a "Good" Wireless Voip Keyset that works well with Asterisk? I would prefer one that is IAX2 as it works better behind a Nat'd Firewall.. But I am reaching out to you guys as you all would know what would work the best :-) ________________________________________________________________ Sent via the WebMail system at kotbh.net

http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even

Hello, I'm new to LDAP. I've read Device class exists (oid 2.5.6.14) in rfc2256. I've heard a Pluggable Device sub-class (a device with a MAC address) also exists though I can't find its OID at the moment. 1. Does any standard class specifically defines IP Phones or SIP hardphones or ATAs or Trunk lines ? What's their OID ? 2. How does one can find by himself if such classes

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Here is Mexico the phone company uses a DSL router from 2Wire which in my opinion is quite bad. I am having problems getting PAP2T adapters connected to Asterisk using these routers. They connect fine but after about 5 minutes I get a message on the Asterisk console that the ATA is unreachable. So far the only way I have found for the ATA to stay connected more than five minutes is to put it in

Over a year ago I saw a discussion about a standalone device which converted a T1 in/out to SIP in/out (over 10/100 LAN). Anyone recall what this device is? (I'm looking for a standalone device - not a PCI card). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070907/7060b23e/attachment.htm

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Asterisk-users, Our two-part tutorial explaining how to use VoIP and Asterisk in Amazon?s Elastic Compute Cloud (EC2) has garnered quite a bit of attention. But due to the time required to complete the many steps needed to get up and running, some of you have asked if it is possible to create a much simpler to install ?pre-built? Asterisk EC2 ?instance.? In short, yes it is. And we?ve

hello, I would to define and unattended process to configure devices which are http-server-enabled, use DHCP but do not use TFTP-DCHP to configure themselves during boot. Has anyone worked on such subject ? I was thinking of something like : populating configuration file from device web pages (rendering this as generic and flexible as possible) writing a script which reads this file and set

For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess

Anyone have any experience with a Japanese local VoIP termination supplier? I've emailed a few companies looking to setup some PSTN to SIP and SIP to PSTN termination, but no luck so far. Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100505/5068aaab/attachment.htm

Hi everybody, As the Asterisk community is getting larger and larger, I was wondering that the features which are provided in Asterisk and are programmed by the open source community under GPL, or GUIs like FreePBX which also come loaded with wonderful features and uses same Asterisk, are they anywhere violating any patent laws? Most of the features work the same way as Nortel, Avaya and other

Has anyone been able to get asterisk 1.6 running under Xen or Amazon EC2? If yes, can you share your experience please? Is it usable in a production environment? How is the sound quality? Am I likely to suffer from latency issues if the extensions are not located in the US? Any pitfall that I should be aware of? Cheers -- Aryan Ameri

Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to

Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. 2) If there is no wireless network, then it

Hi again and thanks I'm using WinNT 4.0 as my network PDC. The Samba server is only a member server. Any idea how to solve this issue? Thanks, Miki > -----Original Message----- > From: Nigel Marsh [SMTP:Nigel.Marsh@first-linux.com] > Sent: Mon, March 29, 2004 11:59 AM > To: samba@lists.samba.org > Subject: Re: [Samba] Windows client using UPPERCASEto connect to >

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten