search for: metaswitch

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...

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch which my Asterisk box replies to. The exchange looks like: <-- SIP read from 172.b.c.d:5060: OPTIONS sip:metaswitch@206.b.c.d:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.b.c.d:5060;rport;branch...

I am setting up VM off Metaswitch due to a problem with Metaswitch VM. I have a couple days to prove this works and I need a little assist please. I am using TRIXBOX 2.6.2.5 and have the Meta SIP trunk up. I have extensions built that can talk to each other. I took a trace on the TRIXBOX that shows when I dial my test phone on Met...

I've been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk that's setup through my local ISPs Metaswitch. I've setup a very simple SIP Peer, which I've played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend insecure=very host=1.2.3.4 context=keysystem nat=yes canreinvite=no username=1235551212 fromuser=12355...

...c/asterisk/asterisk.conf': == Found Running as group '_asterisk' == Parsing '/etc/asterisk/extconfig.conf': == Found Connected to Asterisk 1.6.2.7 currently running on fax1 (pid = 26225) Verbosity is at least 14 -- Executing [9995551212 at default:1] Macro("SIP/metaswitch-00000001", "receivefax,noah.p at bendtel.com" ) in new stack -- Executing [s at macro-receivefax:1] Set("SIP/metaswitch-00000001", "FAXFILE=/var/spool/asterisk-fax/ 1274995265.1.tif") in new stack -- Executing [s at macro-receivefax:2] Set("SIP/met...

I am having difficulties with customer-bound DTMF being very short & clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN <--> Metaswitch <-SIP-> Asterisk <-SIP or IAX2-> CPE Asterisk is running Asterisk 10.4.0 on a CentOS 6.2 VM residing on a CentOS 6.3 KVM host. Asterisk has one network interface connected to the Metaswitch without NAT to place/receive calls from the PSTN, and a separate interface to connect to CPE eq...

Im trying to get my asterisk box to register to a sip provider without much success. here is my console output in asterisk Dec 13 12:57:17 NOTICE[213005]: chan_sip.c:3982 sip_reg_timeout: Registration for 'voicemail.nexband.com@metaswitch.nexband.com' timed out, trying again -- Got SIP response 403 "From: URI not recognized" back from 208.149.73.5 Urgent handler in my sip.conf i have register => voicemail.nexband.com@metaswitch.nexband.com i have authentication turned off on the softswitch and the sip username...

...tics we could get out of Xen? We''re running Xen 4.2.2, dom0 is Fedora Core18 (kernel 3.6.10-4.fc18.x86_64). Guest is RHEL6.4, kernel 2.6.32-358.0.1.el6.x86_64. Hardware is a Dell PowerEdge R620 server with 2 Intel Xeon E5-2690 6-core CPUs. Thanks, Mark Mark Thebridge Software Engineer Metaswitch Networks mark.thebridge@metaswitch.com<mailto:mark.thebridge@metaswitch.com> +44 (0)2083661177 www.metaswitch.com<http://www.metaswitch.com/> _______________________________________________ Xen-users mailing list Xen-users@lists.xen.org http://lists.xen.org/xen-users

...r. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup takes place and the caller can hear about 1-2 seconds of audio before the SIP provider cancels the call and sends back a BYE message. They haven't made any changes on their end (metaswitch). The wierd part is that yesterday I was having the exact opposite problem (outgoing working fine, incoming calls no audio). RTP setup was correct, but * wasn't responding to the RTP packets. Recompiled asterisk with PRI support for the X100P card installed: make && make install libp...

I'm trying to send CallerID info to a MetaSwitch system over a PRI. The MetaSwitch gets the info exactly as it is sent by Asterisk, but I think it might be having trouble with the Hexadecimal b1 that is being sent just before the first character of the CallerID Name. Does anyone know what the significance is of the b1 being sent here? Or, is th...

...es with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new offer by Adhearsion, which then drops the call on trying to re-bridge the two channels. Is there any way to specify that reINVITEs are not to be accepted at the Asterisk level? Thanks, Luca -------------- next part -------------- An HTML at...

Greetings, I have a problem getting Asterisk registered as a UAC against the MetaSwitch call agent, because the customer insists on running it on a NAT'd box. Thus, the Contact: field in the REGISTER request betrays the private IP address of the Asterisk box, but the source IP of the message is a public one. Most registrars don't have a problem with this, including Asterisk....

...CPE device does not support multiple 18x messages in the same call setup. When we receive the 180 we present ring back to the phone, but when we receive the 183 we get confused and stop the ring back tone, but do not open up the early media path for the ring back to be played from the network. In Metaswitch the configuration knob to correct this is ?Superfluous 18x messages?, I don?t know what it takes to configure Asterisk that way. Can anyone help with this. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20...

...1.3.5060: SIP, length: 383 SIP/2.0 403 From: URI not recognized Via: SIP/2.0/UDP 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 From: "asterisk" <sip:asterisk at 10.0.83.61>;tag=as7444eb08 To: <sip:10.0.138.226>;tag=metaswitch+1+0+e288612a Call-ID: 20afd7e40fb31362355eae245dae1fd6 at 10.0.83.61:5060 CSeq: 102 OPTIONS Server: DC-SIP/2.0 Organization: Content-Length: 0 -- Jeremy Kister http://jeremy.kister.net./

I''m working on an application that does extensive database searching. These searches can take a long time, so we have been working on moving the searches to a backgroundrb worker task so we can provide a sexy AJAX progress bar, and populate the search results as they are available. All of this seems to work fine until the size of the search results gets sufficiently large, when we start

Hello-- Looking for a solution. I'm using asterisk HEAD version, from a day or two ago. Trying to register with a Metaswitch voip server via sip. They gave me a userid, and a password. I plug it into a register command in sip.conf: register => 3074449999:pword@isp [isp] realm=voip.isp.net auth=3074449999#c491b58f6fd6da12691fa0de86fbbcc3@voip.isp.net type=peer context=workline host=voip.isp.net username=3074449999...

I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register against a Sylantro server in front of a Metaswitch. I'm able to register and receive inbound calls but outbound calls are rejected by the far end. The username and password have been checked repeatedly. Putting the same authentication and server IP into a softphone or polycom phone work fine for inbound and outbound calls. Has anyone made...

...mber have the following TO header: From: "Unavailable" <sip:asterisk at server.com>;tag=as120a1079. Don't tell anyone, but we are trying to put on a "We're big enough to own the pricey softswitch" look. Even though I would pick a OpenSIPS + Asterisk combo over a Metaswitch any day. Three words "Service Licence Agreement". Anyhow long story short, is there any way to change the "asterisk" part only for the calls that are private? Everything else can stay the same.. Kind Regards, Nick. -------------- next part -------------- An HTML attachment wa...

...re a major carrier. They are also a major pain to deal with, and I'm not sure they'd be willing to sell a VOIP package. Hmmm.... if our target is to manage a maximum of 2000 concurrent calls (for arguement sake), I suppose a softswitch is overkill. Although I did notice that companies like MetaSwitch operate as small as 400 lines. So if our only option was a TDM based solution, and we used Asterisk as our "softswitch", what signaling would be ideal to keep the cost at a minimum? I don't recall Asterisk supporting SS7. Besides I read the Telco's only allow SS7 products they ha...

...-less the same thing without the delay exten => MWISend,2,MWISend(${MWISEND_DIGITS}) ; the variable set in the AMI call - this is tested & works. The Originate goes like: Channel: Local/MWISend at default Extension: 650nnnnnnn Context: direct_out ; a context that dials SIP/${EXTEN}@metaswitch which is known to work Priority: 1 What happens is the local channel goes immediately into Hangup with many odd-looking messages about searching for the extension, when it has clearly already been found and goes into a hangup(see trace below). Anybody have a hint what's happening here and ho...