[asterisk-users] ASterisk and SER
HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk and SER
On 04/12/06, Arun Kumar [EMAIL PROTECTED] wrote: HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. Your dialplan. (Since you didn't get around to posting any configuration or log information, that's about as close as anyone's going to get to your problem). Peter -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASterisk and SER
What is the purpose of that sort of call routing, it does seem like a loop to me. Asterisk is probably getting re-invited to itself... On 12/4/06, Arun Kumar [EMAIL PROTECTED] wrote: HI, My Asterisk is registed with my SER. My client are connected to asterisk when they dial any no like 6 asterisk passes this is ser and then again ser passes this no (strip 1) back to my asterisk. but insted of ringing this exten it says loop detected. can some one tell me what is wrong. thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with SER
HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with SER
Have a look at the OpenSER and Asterisk part of http://openser.org/dokuwiki/doku.php and http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER Arun Kumar wrote: HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SER
On Tue June 20 2006 08:23, Daniel Salama [EMAIL PROTECTED] wrote: I have been reading about integrating Asterisk with SER to help Asterisk deal with large volume of registrations (mainly). I was planning on fronting Asterisk with SER for that purpose. Not that I have the traffic at this moment, but because I wanted to get the infrastructure in place. However, my providers are using G711 codec and I offer G711 and G729 to my clients because they don't have the best broadband service available. So, if my clients are talking G729, I suppose I will have to always keep Asterisk in the media path so as to do codec translation. Is that correct? I was also planning on using SER's nathelper, but if Asterisk _HAS_ to be in the media path, there may not be a need for SER's nathelper. Is this assumption correct? If my purpose of using SER is basically to alleviate registration load and help route (possibly load balance) traffic among multiple Asterisk servers as well as SIP providers, do I really need SER? Would you recommend it? Granted, I have been running both Asterisk and SER as separate systems for a while and they both seem very stable to me. Thanks, Daniel Hi Daniel. How many registrations are you able to achieve per Asterisk server at the moment? As you pointed out, one huge benefit of SER is that it can handle more registrations than Asterisk. We're talking in the thousands here for SER. Again, as you said, if one of your customers elects to use G.729, their call will have to go through Asterisk for transcoding purposes. Note though that if you send G.729-calls through Asterisk and G.711-calls through SER, you'll have to do some extra work to combine the CDRs. Also, you'll have to ensure that the Asterisk and SER servers have the exact same time (IE: use NTP). Having dedicated SER boxes can be very effective at reducing the load on Asterisk servers. Before implementation though, you have to make sure that you have solutions for the wrinkles this modification puts into your original design. Cheers, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
I have been reading about integrating Asterisk with SER to help Asterisk deal with large volume of registrations (mainly). I was planning on fronting Asterisk with SER for that purpose. Not that I have the traffic at this moment, but because I wanted to get the infrastructure in place. However, my providers are using G711 codec and I offer G711 and G729 to my clients because they don't have the best broadband service available. So, if my clients are talking G729, I suppose I will have to always keep Asterisk in the media path so as to do codec translation. Is that correct? I was also planning on using SER's nathelper, but if Asterisk _HAS_ to be in the media path, there may not be a need for SER's nathelper. Is this assumption correct? If my purpose of using SER is basically to alleviate registration load and help route (possibly load balance) traffic among multiple Asterisk servers as well as SIP providers, do I really need SER? Would you recommend it? Granted, I have been running both Asterisk and SER as separate systems for a while and they both seem very stable to me. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
Hi all I have SER installed and running But ser send all voice message to email But i would like to integrate with Asterisk IVR in the like , did some one integrated this kind of setup if so kindly guide me how can i do that ram ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER hangup issue
Hi I have can get my phones to register with SER and dialout for PSTN via my Asterisk box over a SIP channel to my VoIP provider. If the phone requests hangup then the bridged channel on Asterisk gets destroyed however if the called party hangups the channel stays up and the phone connected. Anybody got any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk or ser
hi, SER is less about the number of callers than it is about the number of registered sip clients. Without NAT issues a pizza box server with SER can essentially register an unlimited number of SIP clients. With larger numbers of SIP clients i find SER handles them much better than asterisk. Now, i know this is unorthodox, but i route EVERY call through asterisk, even calls between SER users, for a few reasons: 1. billing - asterisk is much better at keeping CDRs 2. call control - asterisk can stay in the media path if neccesary, SER won't (by default, although you can use a b2bua), and for things like prepay calling cards this is a neccesity. 3. significantly easier to use dialing logic. on the down side, i still havent gotten my old setup with autocreatepeer=yes in my sip.conf and rewritehostport in ser.cfg working on 1.2 - anyone have any ideas about that? hope this helps, -yair On 4/14/06, Xaji Gaid [EMAIL PROTECTED] wrote: Hello:I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic. Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated. -Gaid___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk or ser
Hello:I noticed in few references that asterisk and ser and complementary. Meaning asterisk handles connections to PSTN and voicemail but SER is better for routing SIP traffic.Is anyone using just asterisk for production purpose. Meaning serving a high number of callers. Is it mandatory to use SER behind asterisk? your feedback would appreciated.-Gaid ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
Hello all, I have setup my Asterisk and SER boxes. implementing the ser.cfg and extensions.conf logic i am able to make calls from asterisk to ser and vice versa. is it possible to make simultaneous calls to a ser client from different asterisk clients without getting a 486 busy from SER. Thanks, AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Ser
Hello, when SER redirects calls to asterisk can it be redirected to a realtime peer in asterisk.i cld redirect the call to a static peer but if someone can guide me through for realtime peer settings. Thank you, AA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER for Call Center Application
I suppose the * and SER topic has been discussed way too much, but I searching through all the archives, I haven't really found an answer to what I think could be done. I would like to setup a set of asterisk servers with identical configuration files so that a SER machine can load balance the traffic to, say, 3 asterisk servers. The idea is that if one asterisk server fails, the other servers will take over without having to change any configuration settings. Obviously all the active registrations and channels in the failed server will be lost, but at least the UAs can automatically reregister. The problem that I have with this is that say I have a UA SIP/1001 that registers on Asterisk Server 1. When SER receives a call for SIP/ 1001, all three asterisk servers know about SIP/1001 because it's configured in their sip.conf. However, the UA SIP/1001 is currently logged in asterisk server 1. How does SER know which server to send the actual call to? Could I do something like a broadcast of the call so that all three asterisk servers try to reach the UA and whichever answers answers? What happens if the two asterisk servers which received a request and could not reach the UA send the caller to voicemail, while the call is actually established in server 1 (may be I don't know enough about SIP)? What I'm trying to achieve is the following. In addition to the three asterisk servers, I would setup a central queueing server with asterisk. Say, UA SIP/1001 is registered in asterisk server 1. When the agent on SIP/1001 logs into the queue (possibly via AGI script), the UA will be added to the central queueing server instead of the server the UA is actually registered. Now the central queueing server knows that SIP/1001 is ready to take calls from a queue. Going back to my previous paragraph, when the central queueing server needs to send a call to SIP/1001, it will do so through the SER server. That way, SER can take care of locating (or broadcasting) the call to SIP/ 1001, regardless of which server the agent is actually registered in. This would allow me to have multiple asterisk servers handling all our queue calls. Why would I want to use asterisk for servers 1,2,3 instead of just SER and the single asterisk queueing server? We have many agents in different geographic locations and we need to have all calls recorded. This would allow us to have a distributed architecture of asterisk servers where each server would Monitor each agent's call instead of trying to fine tune so many different details in order to achieve 512 simultaneous calls being recorded. Am I dreaming? Is this conceptually crazy or is it doable? Can someone point me in the right direction? I have some time in my hands and if someone gives me some pointers, I guess I could try to tackle a small lab environment to simulate this. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SER for Call Center Application
It's very doable. I did a presentation of a case study on this exact solution at Astricon last month. Contact me off list for the slides. On 11/3/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I suppose the * and SER topic has been discussed way too much, but I searching through all the archives, I haven't really found an answer to what I think could be done. I would like to setup a set of asterisk servers with identical configuration files so that a SER machine can load balance the traffic to, say, 3 asterisk servers. The idea is that if one asterisk server fails, the other servers will take over without having to change any configuration settings. Obviously all the active registrations and channels in the failed server will be lost, but at least the UAs can automatically reregister. The problem that I have with this is that say I have a UA SIP/1001 that registers on Asterisk Server 1. When SER receives a call for SIP/ 1001, all three asterisk servers know about SIP/1001 because it's configured in their sip.conf. However, the UA SIP/1001 is currently logged in asterisk server 1. How does SER know which server to send the actual call to? Could I do something like a broadcast of the call so that all three asterisk servers try to reach the UA and whichever answers answers? What happens if the two asterisk servers which received a request and could not reach the UA send the caller to voicemail, while the call is actually established in server 1 (may be I don't know enough about SIP)? What I'm trying to achieve is the following. In addition to the three asterisk servers, I would setup a central queueing server with asterisk. Say, UA SIP/1001 is registered in asterisk server 1. When the agent on SIP/1001 logs into the queue (possibly via AGI script), the UA will be added to the central queueing server instead of the server the UA is actually registered. Now the central queueing server knows that SIP/1001 is ready to take calls from a queue. Going back to my previous paragraph, when the central queueing server needs to send a call to SIP/1001, it will do so through the SER server. That way, SER can take care of locating (or broadcasting) the call to SIP/ 1001, regardless of which server the agent is actually registered in. This would allow me to have multiple asterisk servers handling all our queue calls. Why would I want to use asterisk for servers 1,2,3 instead of just SER and the single asterisk queueing server? We have many agents in different geographic locations and we need to have all calls recorded. This would allow us to have a distributed architecture of asterisk servers where each server would Monitor each agent's call instead of trying to fine tune so many different details in order to achieve 512 simultaneous calls being recorded. Am I dreaming? Is this conceptually crazy or is it doable? Can someone point me in the right direction? I have some time in my hands and if someone gives me some pointers, I guess I could try to tackle a small lab environment to simulate this. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk (multiple) + Ser
I have several Asterisk servers installed and one SER server which will act as a gateway to PSTN, en redirect server. I was thinking to implement it the following way: - Register all the * servers at SER (is this neccessary?) - this works via register=asterisk:[EMAIL PROTECTED] in sip.conf - Setup aliases in SER for the telephonenumbers to the appropiate * server: serctl alias add [EMAIL PROTECTED] [EMAIL PROTECTED] e-mailaddress This way, when one SIP phone behind a * server calls for example 016234567, the * server forwards the request to SER, SER looks up the alias en then forwards it to the destined * server. If a number cannot be handled, SER will forward it to the PSTN gateway. Now my problems: I'm a totaly newby on SER. I managed to get the * server register themselves with SER, and setup Aliases. However I cannot get ser.conf configured so that it does what i've explained before. Is anybody willing to help me out, if possible with a sample ser.conf? TIA, Ronald Voermans ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER and Asterisks Queues
Hi all, Can someone help with with Asterisk, SER, and Asterisks Queues? I have three servers: Server A: Asterisk with TE410 connected to PSTN Server B: Asterisk connected to Server A via IAX2 trunk Server C: SER where SIP agents register/connect to What I wanted to do is configure Server A so that it would route certain DIDs to specific UA that are registered in Server C. I don't think this is a big issue from what I've read on the wiki and this list. However, when other DIDs come in, I wish them to be queued in Server B. Server B will then use whatever queuing strategy is defined to target UAs in Server C. Has anyone done this? Will it work efficiently? Also, and I know this is off-topic from my original subject line but I couldn't find it on the wiki, how can Asterisk and SER be configured so that Asterisks handles all voicemail for SER? Can someone point me in the right direction for me to read up on it? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with ser project, , , , here we go! ready or not!!!
I every one, looking for suggestions, or even better yet, a how to guide. I have read most of the wiki's on both, so I know this is completely possible I have Asterisk and SER configured on the same server x6 yes, 6 servers. If I can get this to work on one, we are golden! Number one priority, re-invite = yes number 2 priority = collection of CDR number 3 priority = Asterisk functionality upon request. Ok, I am sure you have all guessed where I am going with this. Here is how it is supposed to work. Ser replicates authentication as a proxy to all servers, to both Asterisk and SER. Harmonic authentication... ( I find many cases of people getting this to work. A traditional outbound call to an Internet based (no Voice T-1 here), hits ser which redirects and collects the cdr and stateless. All MySql databases are nicely mirrored right now.. end user request voice mail and/or conference. Ser forwards request to Asterisk. Asterisk ACKs and does it's job. I am not a leech, I am and open participant in the mailing list already. I am using a number of different systems, mainly FC4 for x64. I have a few Centos/Digium systems and the occasional Red Hat. It is up to me to solve the permissions stuff. Suggestions? Pointers? Brad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk with ser project, , , , here we go! ready or not!!!
Can anyone tell me if the acc_flag is only supported under the options request? one of my telco pipes does not support the options field. Yes, I know, non RFC. Brad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER on Same Box
Does anyone know if it is possible to run both Asterisk and SER on the same box? I am looking to use SER as the SIP proxy while sending SIP calls to a local Asterisk processfor vmail.I am assuming that I would have to change Asterisk from listening on 5060 to some other port to make this work. Is this possible? Are there any other issues? Thanks smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SER on Same Box
Yes it works and as you said all you need to change is the listening port for one of both. Zoa. Keith O'Brien wrote: Does anyone know if it is possible to run both Asterisk and SER on the same box?I am looking to use SER as the SIP proxy while sending SIP calls to a local Asterisk process for vmail.I am assuming that I would have to change Asterisk from listening on 5060 to some other port to make this work. Is this possible? Are there any other issues? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with ser to share the load
Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with ser to share the load
Hello Deepak, 1. don't post multiple times. it's annoying. enough said. 2. run asterisk in verbose mode (start it with asterisk -vgc), place a call from a SIP endpoint behind SER to the asterisk server, and see what happens in the asterisk CLI. 3. if you don't see anything there, get ngrep and place a call from the SIP endpoint while running ngrep SIP and post the output. 4. are asterisk and SER on the same machine? 5. if all else fails put autocreatepeer=yes in your sip.conf - this has bad security consequences, but it is useful for debugging. -yair On 12/2/04, Deepak Dhiman [EMAIL PROTECTED] wrote: Hi friends ! Can anybody help me to configure asterisk with ser so that I can share the load of the asterisk with ser server. I have tried it but my asterisk is not showing registrations of the user agent, as given in the asterisk wiki/asterisk+at+large. I don't know what is the problem, but can assure abt the ser that is is running well and also forwarding packets to asterisk server but * is not getting these packets. Can anybody tell me that what`s wrong with my Asterisk server? Do I need to change /add something in sip.conf? Please help me . Regards, Deepak Dhiman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
I'm a little confused between the pros/cons or benefits of one Asterisk or SER. I've been using Asterisk for a little bit and I know it's a very powerful and scalable platform (in terms of capacity and functionality). However, I've read in some posts that some people are using SER + * and I read no SER's page that it can even do what * does. So, if I have an application where I wish to offer telephony services (call origination/termination) with applications such as prepaid services, conferencing, IVRs, ACD, dialers, etc, would a single installation of * be sufficient? By single I mean * alone (it could be a cluster) and not with SER. I guess to some degree it may depend on the number of clients, but imagine trying to offer a service similar to Vonage. I buy a simple SIP phone, bring it home, sign up for service at www.beyourownprovider.com, register my phone on the site and voila - ready to make and/or receive calls. I can call other members, similar to FWD or I can terminate to PSTN. Now, multiply this scenario to hundreds or thousands of SIP phones spread all over the place, in-front and behind NATs. What would be your approach? Would you still use SER for anything? Is this the right list to post this question? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and SER
SER is a SIP proxy, Asterisk is a PBX, and application server. SER passes calls from place to place and does not get in the audio path. SER uses SIP, * is able to transcode, and convert Protocols. You can build an IVR, VM, and PBX with Asterisk. SER is like a traffic cop, where * is the car wash, garage, gas station, etc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, April 21, 2005 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and SER I'm a little confused between the pros/cons or benefits of one Asterisk or SER. I've been using Asterisk for a little bit and I know it's a very powerful and scalable platform (in terms of capacity and functionality). However, I've read in some posts that some people are using SER + * and I read no SER's page that it can even do what * does. So, if I have an application where I wish to offer telephony services (call origination/termination) with applications such as prepaid services, conferencing, IVRs, ACD, dialers, etc, would a single installation of * be sufficient? By single I mean * alone (it could be a cluster) and not with SER. I guess to some degree it may depend on the number of clients, but imagine trying to offer a service similar to Vonage. I buy a simple SIP phone, bring it home, sign up for service at www.beyourownprovider.com, register my phone on the site and voila - ready to make and/or receive calls. I can call other members, similar to FWD or I can terminate to PSTN. Now, multiply this scenario to hundreds or thousands of SIP phones spread all over the place, in-front and behind NATs. What would be your approach? Would you still use SER for anything? Is this the right list to post this question? Thanks, Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
Hi Everyone, Just a curious question. Has anyone heard of any service provider who is using Asterisk and SER to provide their VOIP services? Thanks Walid ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with SER
Hi all, I am trying to configure * to work with SER (Sip Express Router),the configuration that I am trying is as follows. I have 2 windows machines running X-Lite soft phonesThe * registers with the SER,I want a call from one X-Lite to asterisk(after registration) which is to be forwarded to other X-Lite.For now, I am running both SER and * on the same machine (IP:10.232.2.249) with * on port 5061 and SER on 5060 The contents of the sip.conf is as follows [general]port=5061...context=from-sip...register = asterisk:[EMAIL PROTECTED]:5060/12345 [ser]type=friendusername=asterisksecret=passwordhost=10.232.2.249:5060 [12345]type=friendusername=12345host=dynamicdtmfmode=inband In extensions.conf [from-sip] exten = 12345, 1, Dial(SIP/12345)exten = 12345, 2, Hangup The call is not established, error is: "499: not acceptable here"What can be the problem?. Am I missing something in the configuration? Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term'___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER Integration together
I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive benefits, however, my initial playing around in SERs configuration indicates its NOTHING like Asterisk at all, and almost 5x as difficult to understand and configure. But thats only after a few hours of playing with it. Im interested in learning SER more, especially the integration with Asterisk. Is there a good how-to guide with lots of examples on how to accomplish this optimal setup? Anybody got any good links or resources or can help me with examples? Right now I have Asterisk doing all the work and its getting frustrating w/ Quality issues left and right and such. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SER Integration together
On Wed, 9 Feb 2005 11:44:30 -0500, Paul Rodan [EMAIL PROTECTED] wrote: I know FWD uses a SER/Asterisk combo, and I keep hearing about the massive benefits, however, my initial playing around in SER's configuration indicates it's NOTHING like Asterisk at all, and almost 5x as difficult to understand and configure. But that's only after a few hours of playing with it. I'm interested in learning SER more, especially the integration with Asterisk. Is there a good how-to guide with lots of examples on how to accomplish this optimal setup? Anybody got any good links or resources or can help me with examples? Right now I have Asterisk doing all the work and it's getting frustrating w/ Quality issues left and right and such. It's not nearly as easy as Asterisk is to understand, but it works. I have really very limited experience with it, but just to give you an example of the support out there for it: I was trying to figure out how to forward a call from a user who called to [EMAIL PROTECTED] to [EMAIL PROTECTED] and I tried many different ways. I couldn't get it right. I read the how-to, read through the document, and it wasn't very clear. I went onto IRC and asked for help, and someone told me that I should just use Asterisk. I said I didn't want to, and gave reasons why. They said that to do a forward it was very complex, and I'd be better off just using Asterisk or hiring a consultant to build it for me, and that they had no idea how to do it. Anyhow, I figured it out finally, after reading a totally unrelated mailing list thread. Fast forward a half hour, this same person asks me how many lines we have at my company and states that they would like some of our business... SER seems powerful and it's all the rage, and from what I can tell, it'll be fun to learn it... I'd like to see better documentation for it, and if I ever get to the point where I can provide it, I will. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER differences
Hello All, I have been trying to do a lot of reading lately on the Asterisk project, and I would like to know of someone could tell me mor about the PRO's and CON's between ASTERISK and SER? I have heard that some people use them both in a particular VoIP design and I would like to try and understand why you might want to do that along with how the two can work totegher and communicate. Any information or comments would be greatly appreciated. Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SER differences
Hi SER is a proxy, asterisk handles the media part much better, if you add them together you get a front end proxy which can carry out all your routing functions and then pass to ur media servers at the back. Asterisk can also terminate the calls to pstn , with ser you will need additional equipment, again add them together and you have a tight solutions. Iqbal On 2/7/2005, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello All, I have been trying to do a lot of reading lately on the Asterisk project, and I would like to know of someone could tell me mor about the PRO's and CON's between ASTERISK and SER? I have heard that some people use them both in a particular VoIP design and I would like to try and understand why you might want to do that along with how the two can work totegher and communicate. Any information or comments would be greatly appreciated. Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER security doubts
Hi, I have configured * with a x100p and a E100 E1 card and everything is working fine, now I have setup a SER which the UA would connect, I will be using the * box as a E1 gateway and Voicemail. Anyway, I was alarmed after I tried the integration, when the SER forwards any call to the PSTN the * box wont check any credentials! Im a newbie so maybe this is the correct behavior for the SIP protocol, but when I try to connect directly to the * box it requires authentication, is there a way to setup * to require authentication? Can SER be registered in *? Any help would be much welcomed to better understand the integration of both softwares. Thanks in advance, Humberto ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk receiving SER calls
Hi
Im trying to make Asterisk receiev SER calls and
then redirect them to GNUGK.
But until now, Asterisk isnt receiving
nothing...
Asterisk is already as a gateway in GNUGK as shown in the gnugk
monitorization:
RCF|Asterisk_ip:Asterisk_port|asterisk:h323_ID=ASTERISK:h323_ID=664:dialedDigits|gateway|8478_endp
I installed oh323 for Asterisk. The versions I have are the
following:
pwlib-1.6.6-0_11.rh9openh323-1.13.5-0_13.rh9gnugk-2.0.8-linux-x86
.
In ser.cfg I have just this:
rewritehostport("Asterisk_ip :
Asterisk_port");
t_relay();
that should be enough.
in
sip.conf I have:
[general]context=defaultautocreatepeer=yescanreinvite=no
[mic-inout]type=friendsecret=**username=asterisk
fromuser=asteriskhost=my.domain.com.pedtmfmode=rfc2833insecure=very
And in extensions.conf I putted
exten =
_00NXX.,1,Dial,OH323/${EXTEN}
What is missing to make Asterisk receive calls from
SER?
Joao Pereira
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[Asterisk-Users] Asterisk and SER
Hi All, Can someone help me clear up some stuff? I am about to implement asterisk for a office of about 20 people. I plan on running SIP phones for everyone. (a mix of Cisco Sets and Xlite soft phones) We will place the Asterisk server at a collocation provider and have in connected to the PSTN via 2 PRIs. (digium card) When customers call our 800 number they will be sent to asterisk. When they enter an extension I want asterisk to check if that SIP users is logged in and if not transfer the call back out over PSTN (to a cell phone) Now, here is where things are a little foggy... I want put a local Asterisk server here in the office so that the SIP users connect to it thereby reducing the chatter across the WAN. I would like to have the two Asterisk servers communicate via IAX. Questions: 1. Does this scenario pass muster? Is my thinking logical or does anyone have a better suggestion? 2. Is this possible? Can the remote Asterisk server check to see if the SIP user is logged in to the local Asterisk server before sending the call across the WAN? 3. Should I be using SER vs. another Asterisk server? The problem I see with this is that it doesn't support IAX. I believe that is the preferred method? Am I right? Thanks for all the help from the OSS community. Great software!!! ~chris Christopher Jacob Eye Street Software Program Manager,14151 Newbrook Drive Partner Products Suite 250 301.305.0991Chantilly, VA 20151 www.eyestreet.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER
Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk canuse 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: 6.0.712 / Virus Database: 468 - Release Date: 6/30/2004
Re: [Asterisk-Users] Asterisk and SER
--- Kurtz [EMAIL PROTECTED] wrote: 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. Asterisk is not a SIP proxy, it is a UAS, and also a SIP Registrar. Many use SER and Asterisk together, SER as SIP proxy and Asterisk as PSTN gateway. The advantage is that this combination is highly scalable. Not sure whether Asterisk supports RADIUS authentication. AFAIK, it is not supported, but i beleive some works is in progress in this direction. Do a search on the archives, and you'll get many links on this. Regards, Girish __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk and SER
Hi, It is good that the SER is used for SIP Proxy. But in this case how to use the PBX capabilities of Asterisk like IVR, VoiceMail, DialPlan, and etc.? Best Regards, Miroslav Nachev --- Kurtz [EMAIL PROTECTED] wrote: 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. Asterisk is not a SIP proxy, it is a UAS, and also a SIP Registrar. Many use SER and Asterisk together, SER as SIP proxy and Asterisk as PSTN gateway. The advantage is that this combination is highly scalable. Not sure whether Asterisk supports RADIUS authentication. AFAIK, it is not supported, but i beleive some works is in progress in this direction. Do a search on the archives, and you'll get many links on this. Regards, Girish __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] Asterisk and SER
Hello, --- Miroslav Nachev [EMAIL PROTECTED] wrote: It is good that the SER is used for SIP Proxy. But in this case how to use the PBX capabilities of Asterisk like IVR, VoiceMail, DialPlan, and etc.? It's not that difficult. You can route calls from SER to Asterisk depending on your requirements. You can do this by adding the necessary routing logics in the SER configuration file. On the Asterisk side build a dialplan to handle such calls for IVR, Voicemail etc. Best Regards, Miroslav Nachev Regards, Girish __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTERISK V. SER
I think I do agree with your assessment that *BSD are more stable that linux, no disrespect meant to linux as I think it is wonderful in its own right. What version of FreeBSD ? BSD you are using ? I am looking to build an athlon 64 server soon and am wondering if FreeBSD would be a better option. Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Aaron J. Angel Sent: 15 June 2004 20:17 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK V. SER Usedcanon wrote: Just out of interest (as I am a freeBSD fan) why more stable on BSD ? I have no idea, it just seems to run better on *BSD. I'm still trying to investigate that myself. Perhaps I'm just inept when it comes to Linux, but it has never run decently for me -- I've always had problems with whichever distro I try. This time it seems to be the network card mostly, but then I get similar response from the console every now and then, so maybe it's not the NIC. Maybe I should have rephrased that or left it out, as it's probably not Asterisk that is more stable, technically speaking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTERISK V. SER
Usedcanon wrote: Just out of interest (as I am a freeBSD fan) why more stable on BSD ? I have no idea, it just seems to run better on *BSD. I'm still trying to investigate that myself. Perhaps I'm just inept when it comes to Linux, but it has never run decently for me -- I've always had problems with whichever distro I try. This time it seems to be the network card mostly, but then I get similar response from the console every now and then, so maybe it's not the NIC. Maybe I should have rephrased that or left it out, as it's probably not Asterisk that is more stable, technically speaking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK V. SER
Getting back to the original questions ASTERISK V SER. From reading on SER it's designed to purely to be a SIP server (registrar, proxy, redirect) and is much better than * at that. Since SER is SIP based it will be handling call control and not voice traffic. As mentioned on wiki SER * work well together is a scalable VOIP solution SER handles registrar (provisioning) and proxy, * is PSTN gway/voicemail. Don't ask me howto this is preaching without practice. The VOIP cookbook has a good overview which you can download from http://www.informatik.uni-bremen.de/~prelle/terena/. Br /Kev/ - Original Message - From: Aaron J. Angel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 15, 2004 8:16 PM Subject: RE: [Asterisk-Users] ASTERISK V. SER Usedcanon wrote: Just out of interest (as I am a freeBSD fan) why more stable on BSD ? I have no idea, it just seems to run better on *BSD. I'm still trying to investigate that myself. Perhaps I'm just inept when it comes to Linux, but it has never run decently for me -- I've always had problems with whichever distro I try. This time it seems to be the network card mostly, but then I get similar response from the console every now and then, so maybe it's not the NIC. Maybe I should have rephrased that or left it out, as it's probably not Asterisk that is more stable, technically speaking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTERISK V. SER
Hi Guys, Can somone explain differances between SER and ASTERISK. I am particularly interested in functionality that is not available with ASTERISK but SER can provide. Best Regards Mit freundlichen Grüssen Meilleures Salutations med vennlig hilsen Reza Kordi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK V. SER
Hello, I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution combining multiple protocols (IAX, H323, SIP, Skinny, MGCP, SCCP) so that they can each talk to eachother and multiple codecs (one can use G729 and the other can use ULAW for example). Asterisk also provides other features such as voicemail, hold on music, call display, etc. - Joshua Colp. - Original Message - From: Reza Kordi [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 10:55 AM Subject: [Asterisk-Users] ASTERISK V. SER Hi Guys, Can somone explain differances between SER and ASTERISK. I am particularly interested in functionality that is not available with ASTERISK but SER can provide. Best Regards Mit freundlichen Grüssen Meilleures Salutations med vennlig hilsen Reza Kordi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTERISK V. SER
Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK V. SER
Well since a person would normally go for usability and asterisk is originally created for Linux, I said it was a Linux PBX Solution. I have nothing against BSD myself, I have a FreeBSD sitting a few feet away from me. - Joshua Colp. - Original Message - From: Aaron J. Angel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 14, 2004 11:24 AM Subject: RE: [Asterisk-Users] ASTERISK V. SER Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTERISK V. SER
Just out of interest (as I am a freeBSD fan) why more stable on BSD ? Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Aaron J. Angel Sent: 14 June 2004 15:24 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ASTERISK V. SER Joshua Colp wrote: I can tell you what asterisk is but as for SER... well, I've never dealt with it. Asterisk is a linux pbx solution Linux PBX solution is such a narrow point of view. Asterisk also runs on *BSD; yes, conferencing and MP3s are a bit sketchy due to lack of timer (for now), but the rest of it is fully functional (and more stable) on *BSD. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SER Setup Questions.
Dear All, I have the following setup. Quad T1's-Asterisk (PBX)-(LAN-DMZ)-SER-(Firewall)-(Internet) | Local US Help Desk (Snom 200') This setup works well. I can pass calls from over the internet to the Asterisk PBX via SER using X-Ten Lit. I have a couple of questions; How do I tell Asterisk to forward all outbound URI calls to the SER proxy? This works for anyone on the ser itself, but what about someone on another system on the internet? Lets say I wanted to call someone at IPTEL.ORG? How do I tell Asterisk to forward calls that are not on it to ser? How do I append the caller ID so that my calls do not appear to come from Asterisk? Thanks and Regards Shad Mortazavi --- Nexus Technical Manager n|m Nexus Management Inc Sydney
RE: [Asterisk-Users] Asterisk and SER Setup Questions.
Hi
Shad,
1. You
configure that in extensions.conf
exten
= _[prefix to forward to SER].,1,Dial(SIP/[EMAIL PROTECTED] SER
IP],10)
and
register your Asterisk to SER in sip.conf
register = asterisk:[EMAIL PROTECTED] SER
IP]/asterisk
2. you
can do that in extensions.conf for example
exten
= _[prefix to forward to SER].,1,SetCallerID([prefix to append to CPA
number]${CALLERIDNUM})
regards,
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Shad
MortazaviSent: Tuesday, June 01, 2004 4:07 AMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk
and SER Setup Questions.
Dear
All,
I have the following
setup.
Quad T1's-Asterisk
(PBX)-(LAN-DMZ)-SER-(Firewall)-(Internet)
|
Local US Help Desk (Snom
200')
This setup works well. I
can pass calls from over the internet to the Asterisk PBX via SER using X-Ten
Lit.
I have a couple of
questions;
How do I tell Asterisk
to forward all outbound URI calls to the SER proxy? This works for anyone on
the ser itself, but what about someone on another system on the internet?
Lets say I wanted to call someone at IPTEL.ORG? How do I tell Asterisk to
forward calls that are not on it to ser?
How do I append the
caller ID so that my calls do not appear to come from Asterisk?
Thanks and
Regards
Shad
Mortazavi
---
Nexus Technical
Manager
n|m Nexus Management Inc
Sydney
[Asterisk-Users] Asterisk and SER - choppy sound with G.729
Hi, We are using Asterisk running on FreeBSD,as IVR / Voicemail for SER.We have redirected certain callsfrom SERto *.On * there is some 'testing' extension. It's simply playing some demo now;-) As long as I use plain G.711 the sound is nice. When I switch toG.729 the sound is choppy, not recognizable. What is going on? Debug shows everything is normal.. I understand that all jingles / sounds are recorded in gsm format. Maybe I should try to convert them to g.729, as * isnot converting it correctly on the fly? ButI don't see anyhigh-CPU-usage when* is doing this anyway... I'm usingCisco ATA, or X-pro softphone, so I am aware of silence-suppression (I have switched it off). Thank You for any help, Arek Bekiersz [EMAIL PROTECTED]
Re: [Asterisk-Users] ASTERISK X SER
listas iPfone wrote: Hi All I´m trying to use asterisk and ser in the same box. When i start ser my phones don´t connect with asterisk anymore. i have two nics in this machine 192.168.0.31/37 I need to set asterisk and ser to listen in diferente adresses or ports? I can use the two softwares at the same time? how? I have many problems with my nat and asterisk and i think ser can help, i read the wiki documentation on it but i´m confuse on how to do it. Somebody can point me in the rigth direction? Thanks! Miklos Just set the SIP channel of asterisk listen on different port. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Ser - NEWBIE
Hi All, I'm looking to integrate Asterisk with SER Sip server, my goal is to set up a voicemail server for offline users. Since I suppose that this is a tipical situation of usage I'm asking if someone can point me to faq, config examples and related documentation. Tnx for any help! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
