similar to: Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension.

Hi all, I have been using asterisk to do such kind of thing, But I must admitt, this is not 100 % conveniant (Mainly because Asterisk isn't a SIP Proxy). I just wanted to know if you knew/used some kind of SBC or packages which would deal both with SIP AND RTP ! SER/OpenSER woulc be a good SIP Proxy ... but then how to deal with RTP ? Any tip, info greatly welcome ! Thanks, JM

Hi, I would like to use Asterisk as VoiceMail system ... the only issue I have is with DTMF recognition. Which mode should I force into sip.conf ( general, only for peer ? ) so that the Voicemail application is understanding password from users ... inband : works, but has some glitch ... not always good ... don't know why. rfc2833 : doesn't seem to work .. info : said to be not working

Hi all, I encounter an annoying problem using Asterisk. I 'm using SIP. I try to register an Asterisk as a SIP end user with another Asterisk. If I put both asterisk in the same local network, no problem to do it. The asterisk end user registered perfectly with the other (let's call it the registrar). Authentication is enable and works fine. The problem occurs when I put the registrar

Hi, I'm using Asterisk (1.2.4) as a voicemail system for our Softswitch. When forwarding a call to Voicemail, here is somehow what the softswitch sends to Asterisk : In INVITE : Vm Phone Number ( to route the call ) In To : Person who has been called ! In From : Person who was calling ! Of course, I need to send the call into the "Called User" Mailbox (Thus To SIP header) ! So

Hi, I 've a proxy on my network where some calls are routed to .... And as well some extensions on my Asterisk Server. What I would like to do is to accept all incoming calls from the proxy, wherever they are coming from or going to ... but, as soon as I receive a call with the same number as one extension defined in Asterisk (but through the Proxy !) , it refuses the call, saying that there

I know asterisk isn't a real SIP proxy and is more of a multi-protocol pbx with limited SIP support, but... ... is it possible if you have a central registration server that handles all of your dialplan routing and several asterisk PSTN gateways that it routes calls to for an outbound SIP conversation using reinvites and NOT have the registrar box try and send ANY RTP traffic back to the

The date format may have been an issue, and I had noticed the whois changed the expiration date later to 2018, but dig with my local and with google dns still comes back with no ip. If you addess the gandi dns servers you do get the ip address?? So not sure why it has updated the other dns servers? I have a dyndns.org account, and had mapped an address to the IP and it works find, so the

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

I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account

Should be mailbox = 7752365815@vpbx-wpti Best Regards, Ben Bawkon --------- Original Message --------- From: Bruce Komito To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] voicemail notify problem on sip extension Sent: 5/19/2004 4:27:51 PM I'm having a problem with the voicemail notify feature. Although I have the voicemail box configured for the sip extension, the

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm using Asterisk@home 2.5, and i've done: 1-Create a SIP extension. 2-Leave there a Voicemail message 3-Remove SIP extension Then I've create another SIP extension but with the same number of the above one. I found imediately a voicemail message in my voicemail box. Is this a bug? Am I doing something wrong? Best regards, Marco Mouta

hello * Problem: User authentications fails, error messages see below at end of Info Part Any hints what i should check, what goes astray ? I have to analyse / unterstand a legacy installation and move it to an more recent samba INFO Config Samba: ---------------------------- Config wrt LDAP NO DOMAIN !! [global] netbios name =

Hi, I am wondering whether I can use Asterisk as SIP/H.323 Signalling Gateway. The setup I envisage looks as follows: H.323 end-point ---------(ETH)--------- Asterisk ---------(ETH)--------- SIP Proxy/Registrar ---------(ETH)--------- SIP end-point (ETH: Ethernet) In principle, Asterisk would just be used to integrate H.323 end-points into a fully SIP-based core-network. Hence, there

Unable to determine IP address from host name "www.hdt-project.org" Getting this today? Not sure what issue is? I paid for the renewal back in 08/04/2016 and and in 2015, so the domain should be current? But the whois seems to show it is expired? Went to the gandi site, and it doesn't show a renewal option or anything? whois hdt-project.org [Querying

Hi, all, I am a beginner of asterisk SIP, now I have 3 pc, one runs asterisk as a SIP proxy, and the other two run softphone(Ubiquity) as User Agents. As below: User Agent <------------> Proxy Server <----------------> User Agent (Ubiquity) Asterisk SIP (Ubiquity) My sip.conf and extensions.conf is as follows: sip.conf [general] port =

Hi guys, I'm new to Asterisk, so I'm hoping someone can guide me :-) Currently, I am having the configuration as follows : PSTN -> Cisco router -> Sip Express Router -> Asterisk Voicemail I'm able to get the part from PSTN to Sip Express Router working, but I can't integrate Asterisk with Sip Express Router (SER). Basically, SER does all the registering and forwarding

I'm trying to configure SSHA512 passwords and when testing discovered that they were not working as expected. At first i was using Centos 6.4 which doesn't have the glibc CRYPT newest functions ($6$salt$pass) so had to rollback to the Dovecot format ({SSHA512.HEX}saltedpassword+salt ) but I'm unable to let dovecot authenticate properly. Some logs and details: Apr 16 02:55:37

Hi If I have ser sending calls to asterisk, is there a way to get a different block called in sip.conf for each call (based on some variable, NOT username, From:), if not and they all hit one block which has contect=abc, then when that context is called/matched in extensions.conf, how can I have diff features for various groups of users. EG lets say I have a large company with 4 departments

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show