search for: sipproxy

...I still get the error: May 17 23:20:27 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to '"Tony Hoyle" <sip:6001@213.208.99.114>;tag=as5c348356' Relevant chunks here of data are: [pipecall] type=peer secret=xxxx username=xxxx host=sipproxy.pipecall.com [6001] type=friend username=6001 secret=xxxx host=dynamic context=inbound-from-local The log looks like: Sip read: INVITE sip:8378@asterisk SIP/2.0 Via: SIP/2.0/UDP 213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A From: Tony Hoyle <sip:6001@asterisk>;ta...

...wering with non-codec capability 0x1(G723) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.148.246;branch=0 Via: SIP/2.0/UDP xxx.xxx.148.242:5060;branch=z9hG4bK2142c11da4177 Record-Route: <sip:005622408196@xxx.xxx.148.246;ftag=2142c11da4;lr=on> From: <sip:5555832351@sipproxy.magenta.cl>;tag=2142c11da4 To: <sip:005622408196@sipproxy.magenta.cl>;tag=as1be17fe7 Call-ID: 21fb7142-05e9-c19e-821d-0002a400f1e9@xxx.xxx.148.242 CSeq: 177 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:005622408196@xxx.xxx.148.232> Con...

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 (N...

...g 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: Dial(SIP/*$CALLERIDNUM*$DNID*@sipproxy.address) but I'm not sure if that will even parse correctly... So: exten => _X,1,Dial(SIP/*$CALLERIDNUM*$DNID*@sipproxy.address) is what I think I need in my extensions.conf in orde...

Hello list. My phone rings, I pick up, and the conversation is terminated. Every time. The setup : Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server --> ITSP Could it be the SIP proxy of my Endian firewall ?? I have 4 accounts on the Grandstream which listen on port 5060 --> 5063. They have a proxy defined namely my Endian firewall. On this SIPproxy I have a port range defined 11500 --> 1160...

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 <mailto:0845xxxxxxx@sipproxy.pipecall.com/1000> however none seem to work, the sip msg states it is unauthorised. A person called Tony Hoyle on this list managed to get pipecall to work for incoming as h...

...so that asterisk does not generate loop detected for the INVITE. 4) The call gets answered by asterisk but the 200 OK message keeps routing between asterisk and opensips and then asterisk times out with ?*Maximum retries exceeded on transmission*? error. Scenario: 1) User ----7000 at sipproxy.com--------? Opensips --------7000 at asterisk.com-----?Asterisk 2) Asterisk ------200ok-----? Opensips---200ok---?User 3) Asterisk waits for extension input and user presses 7010 4) Asterisk------INVITE 7010 at sipproxy.com--------->Opensips<https://remote.novanet.net/owa/r...

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

...ister 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: [voiptalk] type=peer secret=xxxxx username=xxxxxxx host=voiptalk.org [pipecall] type=peer secret=xxxxx username=xxxxx host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to '"Tony Hoyle" &...

...uel -----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? [...] [pipecall] type=peer secret=xxxxx username=xxxxx host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to '"Tony Hoyle" &l...

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 this stage anyway. I see that as a separate issue. I have a number of organisational entities (oe), each of which has their own Internet domain presence (alice.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:

..."sip debug" to debug my sip channel. And I build my sip proxy(5060) and asterisk(5065) on the same machine. I make a call from 1011 to 9999 on sip proxy, sip proxy forwards this call to "0939749001". And this 0939749001 will be sent to asterisk by sip proxy. sip ua(1011) => sipproxy => sip ua 9999 ( call forward 0939749001) || ======> asterisk ===> cisco 5300 ==> 0939749001 (pstn) I can find $EXTEN is equal to 0939749001 ( a mobile phone number ) and my $CALLERIDNUM is 1011 But how can I get the value of "99...

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

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

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