search for: broadwork

Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using "BroadWorks". The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So they want me to send my SIP responses to the IP:port they send via the Contact:...

I've been struggling with a Trixbox running Asterisk 1.6 for one of our customers as of late. The service provider in question is using BroadWorks and requires a single trunk registration for the trunk group. We have 4 users(lines/numbers) in the TG. The sip trunk is setup as follows: type=peer host=192.168.1.1 fromuser=<tgid> fromdomain=<sip domain> dtmfmode=rfc2833 allow=ulaw insecure=invite,port context=from-trunk secret=...

...NVITE 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 216.26.109.22> Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact: <sip:4144038968 at 147.135.32.221:5060> Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15...

...yes Use as Outbound proxy: yes Manual Override host: host.net What I've determined is that the SIP server (host.net) needs to see 001234@domain.net. >From XTEN successful registration: Authorization: Digest username="001234",realm="BroadWorks",nonce="1111254549745",uri="sip:domain.n et",response="19107e21c530aa0d3efbda6778a3c41e",algorithm=MD5 >From * failed registration Authorization: Digest username="001234@domain.net", realm="BroadWorks", algorithm=MD5, uri="sip:hos...

...de-incoming ;externip = <my static ip> ;domain = internode.on.net,internode-incoming ;fromdomain = sip.internode.on.net ;fromuser = <phone number> ;username = <phone number> ;secret = <secret> ;auth = <phone number>:<secret>@BroadWorks ;insecure = invite,port ;register => <phone number>:<secret>@sip.internode.on.net ;nat = never qualify = yes canreinvite = no ;expire = 240 [sip-out] type = peer host = sip.internode.on.net context = interno...

I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only give me a SIP IP address to configure my asterisk box, when I call them for support or authentication data to load on my sip.conf, they say me that I don?t need such data, so, anyone knows how I would configure my Asterisk box against a Broadsoft peer?

Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>

...:5060;branch=z9hG4bKQFD28QYY8412c000 To: <sip: callphonenumber at companyname.com>;tag=1164309609-1450128136967 From: <sip: didassignedbybluip at companyname.com>;tag=JEDtnWPm Call-ID: kD5sJntD5-0001-@ ipaddress CSeq: 1808 INVITE WWW-Authenticate: DIGEST qop="auth",nonce="BroadWorksXii6gunlzTrqhij5BW",realm="BroadWorks",algorithm=MD5 Content-Length: 0 ACK sip:callphonenumber at bluipaddress:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000 To: <sip:callphonenumbe @companyname.com>;tag=1164309609-1450128136967 From: <sip: dida...

...p:5547371276 at asbw.alestravoip.com;line=qooanvj> Call-ID: BW1622569380212171700499694 at 10.6.30.9 CSeq: 212444374 INVITE Contact: <sip:5567307529 at 200.94.59.150:5060;transport=udp> Supported: 100rel,timer Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Recv-Info: x-broadworks-client-session-info Accept: application/media_control+xml,application/sdp,multipart/mixed Min-SE: 90 Session-Expires: 900;refresher=uac Max-Forwards: 9 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 287 v=0 o=BroadWorks 26026640 1 IN IP4 200.94.59.152...

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

...ip.broadvoice.com>;tag=as6f6dba69 To: <sip:1TTTTTTTTTT@sip.broadvoice.com> Contact: <sip:UUUUUUUUUU@x.x.x.x> Call-ID: 095981b26d97329e4155ccd529617e5c@sip.broadvoice.com CSeq: 103 CANCEL User-Agent: Asterisk PBX Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks", algorithm=MD5, uri="sip:1TTTTTTTTTT@sip.broadvoice.com", nonce="1110785211206", response="f68a31735aec843b9ef68b7909fcf178", opaque="" Content-Length: 0 (no NAT) to 147.135.8.128:5060 Scheduling destruction of call '095981b26d97329e4155ccd5296...

Hey Folks, Is it possible to transfer an incoming call back out without a "trombone" effect. For instance; Caller dials my broadvoice # --> Asterisk Answers and plays a menu --> the caller selects an option --> asterisk transfers the call to my cell phone via broadvoice and removes itself from the equation so I end up with... Caller --> Broadvoice --> Cell Phone Vs.

...23; branch=z9hG4bK68a91703;rport=5060 From: "310XXXYYYY" <sip:310XXXXXXX@sip.broadvoice.com>;tag=as14f78807 To: <sip:8886831000@sip.broadvoice.com>;tag=SD5894199- Call-ID: 213480db053b77d6797973362cb20554@sip.broadvoice.com CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5, nonce="1110838927310" Content-Length: 0 Somewhere the invite is breaking up, but I can't figure out where. It's especially peculiar because it's only on my forwarded calls, normal outgoing calls work just fine. I'm not sure how to fix this, I've...

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

...623304774 INVITE Contact: <sip:4105156666@147.135.0.128:5060;ep=147.135.0.129;transport=udp> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY Supported: 100rel Accept: application/sdp,application/dtmf Max-Forwards: 69 Content-Type: application/sdp Content-Length: 276 v=0 o=BroadWorks 24463992 1 IN IP4 147.135.0.128 s=- c=IN IP4 147.135.0.128 t=0 0 m=audio 14942 RTP/AVP 0 8 2 18 96 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:96 iLBC/8000 a=rtpmap:101 telephone-event/8000 Jul 24 15:24:33 VERBOSE[1078]: 13 headers, 12 line...

...xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" <sip:8475100139@sip.broadvoice.com>;tag=as292b9469 To: <sip:18475098263@sip.broadvoice.com>;tag=SD38ad399- Call-ID: 5e81fefe69a768f969fd50f6000d4a39@xxx.xxx.xxx.xxx CSeq: 102 INVITE WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1110346372627" Content-Length: 0 8 headers, 0 lines Transmitting: ACK sip:18475098263@sip.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: "6050" <sip:8475100139@sip.broadvoice.com>;tag=as292b9469 To: &...

Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ "We're sorry your call cannot be completed at this time" So... I've

I just signed up for the BroadVoice service a few hours ago, but for the life of me I can't get any incoming voice. The incoming connection is fine as it rings my extension from outside, but I can't hear anyone talking. Outgoing voice is working fine though. I've been looking through the archives, but I haven't found a solution to the problem yet. I even tried another router