similar to: Ser vs Asterisk?

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the

> We have our Asterisk server running smoothly with a SIP BRI gateway > for inbound calls. However if the Internet connection goes down and a > DNS server becomes unreachable Asterisk basically does not function. > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the

With all of the recent talk on the list about DUNDi, I have a question. From the outset it appears that SER is often used for high availability solutions and as a tool for almost clustering Asterisk boxes behind it. It appears to me that DUNDi is providing a lot of this as well. Now I know DUNDi is not an application by itself to proxy SIP requests but can I hear any information out there that

Hi all, I have the following setup: UAs ------------SER ------------------------ ASTERISK ---------------------GNUGK --------------- GWs SER is configured to route call requests from UAs to Asterisk. Asterisk is configured to receive the call on SIP channel and dial out to GNUGK over H323 channel. The problem I'm facing is that asterisk sends out the call request to GNUGK and times out

Hi All. I'am using Asterisk with SER. I can make call between two internal VoIP gateways or from na internal to external VoIP gateway. But when I get a external call, this call hang ups 5 seconds after and I reveive the following messages *CLI> -- Executing Dial("SIP/16008-3d17", "SIP/16007&SIP/16006|20|tr") in new stack -- Called 16007 -- Called 16006

Has anyone had problems with using hold on a 7960 SIP firmware? The problem is when the 7960 puts a call on hold and you take it off hold again, the 7960 outbound audio is delayed on the other end. Sometimes up to a few seconds. I've tried a couple different things, making the "other end" a diff type of trunk ie: 7960sip --> asterisk --> IAX2 --> PRI 7960sip -->

Do somebody using CISCO AS5350 with Asterisk? Which protocol do you using: H323, MGCP, SIP? This direction: [12sp->Asterisk->h323->as5350->isdnPSTN] is ok But reverse: [isdnPSTN->as5350->h323->Asterisk->12sp] cannot hear 12sp, but 12sp hear PSTN (codec g711u) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I am trying to figure out which way to go for a quad port fxo solution with a good echo can on it. My options are the sangoma remora, a mediatrix fxo, or something similar. The issue is that I would need a good EC. This would be on about a 9000 foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4 port card. Anyone have experience that drives them in a

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

Hey gang, I'm trying to work out all possible scenarios using SER & Asterisk in our upcomming deployment. The example scenario is 50 different customers, all with different numbers of SIP UAs. All UAs would register with SER; This will help keep any inter-office conversations off our bandwidth since SER doesn't handle the RTP stream. Calls from PSTN to UA are easy to handle.

Hi all, I would like to know if it is possible to use asterisk and ser together in a single computer system using ser as a sip proxy and forwarding any voice call request to asterisk for calling into the pstn gateway. (or any other alternative that is possible is also welcomed for suggestions). If it is possible can someone kindly show me the necessary configuration files or refer me to any page

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

Hi, I'm trying to bundle the powers of Asterisk and SER. Asterisk for pabx functionalities and termination to landline/PSTN, and SER as SIP Gateway/Proxy. With my current configuration the SIP user just adds 0 as a prefix to a number, and the call will go out to PSTN over Asterisk. For this to work I added the rewritehostport() function in SER to point to the Asterisk IP (different from the

Hello, I have several * servers behind a SER server (in a local ip range). The SER server is also publicy reachable. On the other site, I have SIP clients that are behind another NAT or in the same NAT range as the * server. Can someone give me some directions/hints etc. on how to make this work. I think I should be using MediaProxy with SER. But do the SIP clients need to register at the SER

Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version:

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >