similar to: Call ends when picked up

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0

I've been looking through the examples and docs, but haven't yet quite figured out how to do what I want. We've got a T1 coming in carrying a block of telephone numbers, terminating on an Asterisk box. Any call that comes in needs to get sent to a SIP proxy, with a particular extension format: *ANI*DNIS*@sipproxy.address The closest I can see to do this with the Dial() command is:

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

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000

How do I get the bit like "IAX2/white_phone" in extensions.conf eg from pre-defined variables or variants thereof ? What I *do* get is strings like this "IAX2/white_phone@white_phone-4" from ${CHANNEL}, but that's the full channel name. What am I missing here ? Thanks, Thomas

I think I have got * worked out as far as getting users on a small private network talking with each other, but when it comes to the bigger picture about talking between private networks connected by the Internet then I am getting confused about STUN, SER, SIPPROXY, RTPPROXY, etc. Before I start let me make it clear that I am not looking to drop out onto the public telco network anywhere, not at

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to use this account with AsteriskNOW to begin my experience with a VoIP based PBX. The

hi i got TOS and retranssmission error on receiving SIP call chan_sip.c:2794 retrans_pkt: Maximum retries exceeded on transmission 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 for seqno 43156 (Critical Response) -- See doc/sip-retransmit.txt. [May 22 13:42:44] WARNING[18021]: chan_sip.c:2821 retrans_pkt: Hanging up call 10CAED68-0F1D-DF82-DA1E-A76C1CB3D8A3 at 172.18.100.72 - no reply to

*** If anyone has a better way of doing this, please post to the list. I hadn't seen anything on this list or in channel.c/chan_local.c - which prompted this email *** I'm not sure how many VoIP providers out there are using Asterisk as a service platform like we do, but I thought I'd share an experience with call looping that was racking my brain with the list. One of the

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install