similar to: Load balancing Asterisk.

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hi, Does anyone have some good examples of a Kamalio or OpenSips configuration that integrates with Asterisk? Essentially I want to use the SIP router as the UA, but still run all the calls through Asterisk (for dialplan, etc..) I've looked for examples on the project web sites, but I haven't found anything decent yet. Thanks. -- James

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?

I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently I"m using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

Greetings, As a developer and consultant who spends considerable time on projects involving the fusion of Asterisk and products derived from the SER ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have found that there is a great volume of interest in this topic on the mailing lists associated with all communities involved, but a comparative lack of focus that results in

Inspired by AG Projects' Adrian Georgescu's post of Eric S. Raymond's classic "How to Ask Questions the Smart Way" to the OpenSIPS-users mailing list[1], I'm going to repost it here: http://www.catb.org/~esr/faqs/smart-questions.html As Adrian said, "This a good read for those who show up on mailing lists without any guidance about how to ask the right

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Hi, Is it possible to have MOH in early audio, while waiting for someone to pick up a Dial() call? (When using zap channels, I have early audio working with playback) H?kon Nessj?en Loopback Systems AS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070429/c901ff90/attachment.htm

Hello VoIP guys Sorry for being somewhat off-topic. At the moment I am studying informatics in the seventh semester and I need to start thinking about my thesis. As I am very interested in VoIP technologies I thought about picking this as my main topic. So far I have only little experience in this area. I have been fiddling around with siproxd and pfSense and have red the one or the other packet

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying,

Sorry if this hit the list twice, sent out yesterday, but didn't see it show up. Hi All, Can Asterisk be used as a SIP proxy, blah, blah, blah??? I've glanced over questions like this through the years, with a good idea on what a SIP proxy is and what Asterisk is and IS NOT. I never really took the time to lab-up SER and test drive it to see what advantages might be gained from using