similar to: Voxalot User and Peer details.

All, I'm hoping someone can direct me as to why when someone calls my DID Asterisk tries to authenticate the incoming call on my outbound context. If I remove the GoTalk context I can receive incoming calls. Outbound calls work fine while I have the GoTalk context in place. The error I am getting when someone calls the DID is WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,

Hi, I too have found this matching to be frustrating. I would like it to behave as you describe. Doug -- Doug Meredith 506-854-7997 ext. 801 ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep Sent: Thursday, January 04, 2007 1:50 AM To: Asterisk Users Mailing List - Non-Commercial

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

Hi all, I'm running Asterisk SVN-trunk-r8643M and face following problem: I'm trying to get incoming call from a provider and calls ended with a 404 error. On the INVITE I get "Found no matching peer or user for <IP address>:5060" and then "Looking for <UserName> in <SIP default context> (domain xxx.xxx.xxx.xxx)". My question is why asterisk

Hello I've done some googling, but still puzzled at my working configuration. Apparently, a "user" can only receive calls through Asterisk, a "peer" can only make calls, and a "friend" can do both. If that's correct, I don't understand why my VOSP requires the following settings in sip.conf to let my Asterisk server make/receive calls to/from the PSTN:

I'm looking at setting type=peer vs type=user (in both IAX and SIP conf entries), and I found a comment attributed to digium (http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that type=user is depricated and that we should only use type=peer Is that still correct? Will type=user be phased out, and should even new installs of older asterisk versions (eg: 1.6) use

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

Hi All, I have installed Asterisk 1.4.2 and have loaded ztdummy as I have no Digium cards. The problem I have is that MOH will not play. It starts and then stops. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 I'm not sure if the above is correct.

When I do a "sip show peer <peer> load" command in the Asterisk CLI I get the information about the peer I requested, however, there is a line that says "Realtime peer: No". All the other information is correct. According to "help sip show peer" the "Option "load" forces lookup of peer in realtime storage.". Also, this particular peer is

Hi! I wonder why users.conf generates a SIP user and a SIP peer? Why is it not possible to set type=... in users.conf? (Asterisk 1.4.22) thanks klaus

Deer experts, I'm using glusterfs 3.2.5 and I got a cluster of 6 peer. Now one of the peer says all the other 5 peers are in Peer Rejected (Connected) status and the other 5 peers say that peer is in Peer Rejected (Connected) status. And I noticed the if I create a new volume in the cluster the fault peer won't see the volume. Any one known how to recover the fault peer? Thank you very

Hi, I have a little situation here :( I have an Asterisk working, and in another office, a Sipura SPA-3000. I configured the SPA and I have the extension working, the incomming trunk working, but the outgoing trunk (peer) does not work. The issue is that I have a dynamic IP where the SPA is, and neither the SPA nor my router have DynamicDNS. So, if I

Hi, I'm trying to create a link between two PBXs. One is Asterisk 1.4.15, the other is an unknown 3rd party PBX. In my internal testing, beween two A*k servers, I found that if I created two sip accounts from the same IP, one as peer and one as user (intending to give an -IN and -OUT setup), then inbound calls always seemed to route via the -OUT account and failed. My fix was to use

Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes

Can anybody please answer this. Both clients are behind different NAT's. One of them starts a SIP call to the other through Asterisk. Asterisk sets up the call. Issues reinvite and connects them together. After this point does the media stream flow through Asterisk or Peer to Peer? Does such a call use any system resources of Asterisk server after connection? Thank you in advance.

I get this warning in the Asterisk CLI once in a while, and it usually corresponds with a phone not ringing when it should. Warning in CLI: Inringing for peer [PEER] < 0 What does it mean and what is the likely cause of this? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Just to bug you all (feel free to rant at me), a client wants to set his caller*ID number for outbound calls though us to PSTN. the client is using SIP to us, he can set the caller*ID name fine. if he sets his caller*ID number to anything other than his account number (8440101), the call is dropped into the default context (and then hung up by our dial plan). To get around this i

Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten => 500,n,Dial(SIP/romi at 192.168.1.79) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in