similar to: asterisk or ser

Hello all! I googled lists.digium.com and ser mailing list, but did not find any working configuration of asterisk used as voicemail for SER. This is my config if (uri==myself) { if (method=="REGISTER") { save("location"); log (1, "Registered\n"); break; };

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

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

I have some simple questions and i need your help guys. I have ser server which working fine, between users. I am trying to add some more features to the ser. Most important is the IVR. I installed Asterisk and i am trying to register user in asterisk with no success. Part of ser.cfg file where i am trying to redirect the call to the asterisk.

hello all, i have a problem, and i'm tearing my hair out...any assistance is appreciated. I am trying to redirect from SER to Asterisk, both on the same machine. In 1.09 I didnt need to set up a peer for SER, just autocreatepeer=yes, and rewritehostport from SER as below, and asterisk accepted the requests without a problem. When I updated to 1.23 requests from SER to asterisk die quietly, no

Here are some additional informations to the earlier thread "MYSQL - Was Functional till Update to 3.13". It stopped working somewhere between 3.0.12rc1 and 3.0.13. When adding with pdbedit like: pdbedit -a fuman -f "full name" and then after the passwd prompt, it always fails and says "Username not found!" After revert back to 2.0.12rc1 again, it correctly prints

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a

Hi All, I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on 5060 and asterisk on 5065. The setup is that people use serweb to create an account and register a phone. Their calls are routed from ser to asterisk and then inbound on IAX2. The server has a public and an internal interface. The real FQDN of the server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and

Hi, We are testing a SIP solution * + ser solution for a large implementation. All the clients are nated. When a client is dialing outside the domain (to a FWD sip account for example) all is perfect ! ;-) But ,when a call is done to a sip account, the client is ringing, then the caller can hear the nated client very well, but the nated client does'nt hear anything. RTP issue no ? I've

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)

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

We have working in a rural project and I have selected Asterisk. Because * will work with a prepay system we need a precise answer supervision for outgoing calls. The pilot system will have 8 trunks to PSTN. The Telephone Company provides EURO BRI and analog CO trunks. On CO trunks the answer supervision is reversal polarity. The village to be served will be connected through a WLAN. Thus,

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear I am using ser + asterisk, for setting up land line calling. only probelm, each unregistered soft phone can places the call only with callerid, this is critical problem, because any number(soft phone) , has a limit time to use this system, best Mani ____________________________________________________________________________________Be a better Globetrotter. Get better travel

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1">Hi,<br> <br> i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull

Dear R People: Do any of the linear model or regression function calculate the Durbin-Watson test statistic for serial correlation, please? I found the hat matrix, studentized residuals, and so on, but no D-W. Thanks so much! Sincerely, Erin M. Hodgess, Ph.D. Associate Professor Department of Computer and Mathematical Sciences University of Houston - Downtown One Main Street Houston, TX 77002

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I have been working on supporting the TrippLite Internet350SER (serial connection only using communication protocol 1004) in nuts.2.4.1. I sent a patch around 2009/11/9. I just look at the 2.4.3 release and the support for this model does not look to be added in either tripplite driver. Can I help to add support in the project? -- Pat This electronic message may contain proprietary and