search for: sip_proxy

sip.conf.sample: ;register => 2345:password at sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. sip.conf: [general] context=default allowoverlap=no u...

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register => 2345:password@sip.messagenet.it:5061 but I need the other syntax 'ca...

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know the syntax for that. I tried " exten => 511,1,Transfer(SIP/sip_proxy-out...

...to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing Ringing("H323/ip$192.219.85.57:2712/8570", "") in new stack -- Executing Dial("H323/ip$192.219.85.57:2712/8570", "sip/280@sip_proxy-out|20|r") in new stack -- Called 280@sip_proxy-out -- Got SIP response 482 "Loop Detected" back from 192.219.85.57 -- Now forwarding H323/ip$192.219.85.57:2712/8570 to 'Local/280@sip-incoming' (thanks to SIP/sip_proxy-out-f67d) Jul 22 20:20:25 NOTICE[29756]: chan...

...y sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No CallID register => user:password@sipprovider.com [sip_proxy] type=friend username=user fromuser=user secret=password host=siprovider dtmfmode=inband The problem is: If i put in the [sip_proxy] section type=friend, incoming calls doesn't works. If the type is set to another value (for example peer) incoming calls works fine, but outgoing calls doesn...

...preciated. -DevilFish Asterisk Version: Asterisk CVS-v1-0-01/13/05 Call Start: Jan 15 12:46:59 VERBOSE[4290]: -- Executing SetGroup("SIP/302-928e", "302") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- Executing Dial("SIP/302-928e", "SIP/12699264242@sip_proxy-out|30") in new stack Jan 15 12:46:59 VERBOSE[4290]: -- SIP/sip_proxy-out-f201 is making progress passing it to SIP/302-928e Jan 15 12:47:08 VERBOSE[4290]: -- SIP/sip_proxy-out-f201 answered SIP/302-928e Jan 15 12:47:08 VERBOSE[4290]: -- Attempting native bridge of SIP/302-928e and...

...ioned. zapata.conf: [channels] language=en context=zap signalling=pri_cpe switchtype=euroisdn pridialplan=unknown callerid= usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=yes musiconhold=default group = 3 channel => 63-77,79-93 sip.conf: disallow=all allow=alaw allow=ulaw [sip_proxy] type=friend context=foobar host=sip.proxy.net defualtip=x.x.x.x port=5060 disallow=all allow=ulaw canreinvite=no nat=no extensions.conf: [sip] exten => _X.,1,Dial(Zap/g2/01234567890) [zap] exten => _X.,1,Dial(SIP/1234@sip_proxy) Asterisk CLI: -- Accepting call from '1234' to &...

...to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk: -- Executing Ringing("H323/ip$192.219.85.57:2712/8570", "") in new stack -- Executing Dial("H323/ip$192.219.85.57:2712/8570", "sip/280@sip_proxy-out|20|r") in new stack -- Called 280@sip_proxy-out -- Got SIP response 482 "Loop Detected" back from 192.219.85.57 -- Now forwarding H323/ip$192.219.85.57:2712/8570 to 'Local/280@sip-incoming' (thanks to SIP/sip_proxy-out-f67d) Jul 22 20:20:25 NOTICE[29756]: chan...

What is the relationship between the peer definitions and the register command? In reviewing the sample sip.conf it seems that you can place the "sip_proxY" peer as the hostname. Is this correct? This question adds the the Broadvoice thread and where to place the dtmfmode variable. sip.conf --- (asterisk sample) -------------------------------- ;register => 2345:password@sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_p...

Hello, I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial number which is connect to Quintum, and call is diverted to *. I don't know what I should set, if I want call from SIP_phone registred in Asterisk to PSTN via Quitnum. I set in sip.conf account for Quintum [sip_proxy-out] type=peer outboundproxy=QUINTUM_IP , and changed extensions.conf. When I call from SIP Phone, I see in Quintum log, that call is received with good caller and called numbers, but I think that quintum don't how route this call (he diverte this call to asterisk). So, can y...

Hello! I've been struggling with the documentation for months on this simple subject... I still have not been able to get this concept down... I have 3 sip accounts (PSTN DID's) that come into my asterisk box and give me phone service from my itsp via SIP. I for the life of me have not been able to figure out how to get them to come in to 3 seperate contexts! This must be simple but I

...ake an outgoing call > from Asterisk to SER I see the following in Asterisk: > sip debug SIP Debugging Enabled -- Executing Ringing("H323/ip$192.219.85.57:2488/23038", "") in new stack -- Executing Dial("H323/ip$192.219.85.57:2488/23038", "sip/290@sip_proxy-out|20|r") in new stack We're at 192.219.85.57 port 15916 Answering/Requesting with root capability 0x4 (ulaw) 12 headers, 8 lines Reliably Transmitting: INVITE sip:290@fedcore2.eicon.com SIP/2.0 Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805 From: "223" <sip:Aster...

...ccept calls ; from this SIP proxy (provider) ; ; host is either a host name defined in DNS or the name of a ; section defined below. ; ; Examples: ; ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local ; extension 1234 in extensions.conf default context, unless you define ; unless you configure a [sip_proxy] section below, and configure a context. ; Tip 1: Avoid assigning hostname to...

...n one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling number. My dialplan looks like: [globals] ... TRUNK=SIP/sip_proxy-out CELL=${TRUNK}/208xxxyyyy PHILIP=SIP/bedroom_1&SIP/office_2&SIP/kitchen_1&${CELL} [incoming] exten => s,1,Answer() ; sometimes signaling and media get out of sync on cell networks... exten => s,n,Wait(0.75) exten => s,n,Playback(main-menu) exten => s,n(exten),Background(...

...bugs) ;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ; ; Note that promiscredir when redirects are made to the ; ; local; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local ; extension 1234 in extensions.conf default context, unless you define ; unless you configure a [sip_proxy] section below, and configure a context. ; Tip 1: Avoid assigning hostn...

...f file is eating my lunch. Here are my various config files - maybe someone will take pitty on me and point me in the right direction. Needless to say, Asterisk pukes on my dialplan when I try and startup. . (zapata.conf) context=analog signalling=fxs_ks language=en channel => 1 (sip.conf) [sip_proxy] For incoming calls only. Example: FWD (Free World Dialup) type=user context=sip [xlite1] "Transmit Silence"=YES type=friend regexten=1234 ; When they register, create extension 1234 username=xlite1 callerid="Jane Smith" <5678> host=dynamic allow=ulaw allo...

...accept calls ; from this SIP proxy (provider) ; ; host is either a host name defined in DNS or the name of a ; section defined below. ; ; Examples: ; ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ; ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local ; extension 1234 in extensions.conf default context, unless you define ; unless you configure a [sip_proxy] section below, and configure a context. ; Tip 1: Avoid assigning hostnam...

Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

...host is either a host name defined in DNS or the name of a > ; section defined below. > ; > ; Examples: > ; > ;register => 1234:password@mysipprovider.com > ; > ; This will pass incoming calls to the 's' extension > ; > ; > ;register => 2345:password@sip_proxy/1234 > ; > ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider connect to local > ; extension 1234 in extensions.conf default context, unless you define > ; unless you configure a [sip_proxy] section below, and configure a context. > ; Tip 1...