similar to: Problem with incoming calls

When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '<sip:201@192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip ; Default context for incoming calls

Hello All, Some new security stuff is going on I suppose in 1.8 that I am not familiar with and would appreciate your help In a scenario such as the following: Internet --> SBC --> Asterisk upon trying to register an endpoint, the following is being observed on the Asterisk Console. Have Googled this but haven't come up with anything that helped much. [Mar 10 11:53:59] ERROR[21272]:

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hi I have a linux mdk 8.0 w/ custom 2.4.19-16 kernel, acls, quotas etc I have a samba 2.2.7 serving several shares on LVs everything's fine but with one share : whenever a windows client accesses this share his file manager is locked out for exactly one minute. this happens when the client requests the directory listing (or when an automatic refresh takes place) i can't see any reason

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hey all, I'm trying to create redial on busy for my users, but haven't the foggiest on how to make asterisk route depending on the status code returned over SIP (483, Busy Here?). . . anyone know how to do this? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to see if any of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin for a VOIP provider, and have encountered a few PBX customers that want consulting/support for the IAreaNet provided Asterisk pbxs. These guys are selling AAH servers to the public, and are (at least the techs that I talked to trying to get our service working) not knowledgeable at all.

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer

Hi I'm struggling with setting up SIP domains. If I specify a domain and a context in [general], that context overrides any set in type=peer blocks elsewhere. This results in incoming calls from PSTN gateways I use arriving in the wrong context. If I don't specify a context (which the docs I've found suggest is valid), then I get: 2006-05-12 07:36:16 WARNING[95290]:

I did an ngrep trace of what Asterisk realtime static does when it queries the database. Here's what I saw SELECT category, var_name, var_val, cat_metric FROM rt_pbx1_sip_vw WHERE filename='sip.conf' and commented=0 ORDER BY filename, cat_metric desc, var_metric asc, category, var_name, var_val, id; Firstly, why does it order in DESCENDING order by cat_metric? Shouldn't it be

Hi. I'm trying to set up a bunch of SIP phones to register in various domains on the local subnets connected to an Asterisk appliance, yet be able to support Internet SIP calling. My config looks like this: [general] context=INVALID allowguest=yes autodomain=no ... domain=redfish-solutions.com,redfish-internet ... [internal](!) type=friend qualify=50 nat=yes host=dynamic canreinvite=no

Hello list, I would like to seek your expert opinion on a setup I am trying as part of my research. I have not been able to successfully make a call so far. In my setup, I use two laptops that are interconnected by means of a stand-alone IS1581 switch. Thus there is no LAN involved. I have assigned static IPs to the two laptops, say 10.0.0.1 and 10.0.0.2. I have installed Asterisk 1.6.2.6 and

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex -- There is no instance of a country having benefited from prolonged warfare -- Sun Tzu - The Art of War -------------- next part -------------- A

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes

Tzafrir, How did you set sip:tzafrir@local.xorcom.com I use ser----asterisk look at my sip.conf and extensions.conf Regards Harry //////////////////////////////////////////////////// [general] context=sip realm=nxs.yi.org bindport=5050 bindaddr=nxs.yi.org srvlookup=yes tos=lowdelay maxexpirey=3600 defaultexpirey=1000 allow=all musicclass=default language=fr insecure=very allowguest=yes