search for: sip_provider

Displaying 10 results from an estimated 10 matches for "sip_provider".

2006 May 26
0
SIP call problem
Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 The...
2005 Jul 01
2
Sip.conf problems
Hi, I have been trying to configure my Asterisk to use a Sip provider for out and incoming calls. I only have one user and password for connect to my 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
2011 Jan 26
0
SIP channel status - Why is it different when calling an internal extension rather than an outside line over SIP?
...ck line right away. What is the fault behind this and what is workaround? This works: *originate sip/101 extension s at dial_wait* [dial_wait] exten => s,1,Answer exten => s,n,Playback(Please_wait_as_dial_the_second_party) exten => s,n,NoOp(Calling second party) exten => s,n,Dial(SIP/sip_provider/12145556666) This doesn't wait for channel to come up and jumps to Playback (s,2) without even the first party yet picking up: *originate SIP/sip_provider/12148889999 extension s at dial_wait* * * *Thanks,* -------------- next part -------------- An HTML attachment was scrubbed... URL: <ht...
2008 Dec 15
3
Variables for dial plan
...p.conf [client1] dialplan=NZ .......... In extensions.conf (Logic expressed using PHP style) if ($dialplan == NZ) { $NAT = 0; $INT = 00; }; and in the [outgoing] section ; Australia exten => _${INT}61[278]NXXXXXX.,1,Set(CDR(UserField)=AUSTRALIA) exten => _${INT}61[278]NXXXXXX.,n,Dial(SIP/SIP_PROVIDER/0${EXTEN:4:9}) How can I implement this in Asterisk style? Thanks, Michael
2006 Apr 27
1
asterisk spandsp and txfax
...2email on my asterisk box. The rxfax works fine in my setup. The problem is with the txfax. I have tryed all snadsp version (0.0.2x and 0.0.3x) but I get this errors. Because I can't find anything on Internet I'm hoping u can give me a hand. here are my logs: -- Attempting call on SIP/sip_provider/1234 for application txfax(/va r/spool/asterisk/fax/215690048.1145968036.383.tif|caller|debug) (Retry 1) > Channel SIP/mc3810-a20f was answered. > Launching txfax(/var/spool/asterisk/fax/215690048.1145968036.383.tif|ca ller|debug) on SIP/mc3810-a20f FLOW Slow carrier up FLOW Slo...
2006 Mar 07
3
Re: [asterisk-dev] Is there a way to define an outbound proxy in sip.conf ?
...back to the sip proxy. Is it possible ? ASTERISK | | | | ==SIP proxy===sip agents Thanks for help Harry --- Wilmar Campos <wilmar.campos@gmail.com> a ?crit : > Yes: > > > ; > ; Provider or Remote PEER > ; > register => 800:345698:12345678@sip_provider/800 > > [sip_provider] > type=peer > context=default > ;secret=345698 > fromuser=800 > host=sip.provider.com > ;language=es > dtmfmode=rfc2833 > disallow=all > allow=g729 > canreinvite=no > I hope this is what you are looking for. > > Regards, > &g...
2004 Jul 12
0
GnuGK + Asterisk + SIP Provider
...| H.323 SIP And I wanna configure a setup that the SIP terminal talk with the H323 terminal. For this I would like use the asterisk. My h323.conf file is like: [general] gatekeeper=10.11.2.80 AllowGKRouted=yes [H323Asterisk] type=h323 context=sip_provider prefix=113151,116462 [default] type=h323 context=default My sip.conf file is like: [default] context=default My extension file is like: exten=_1131517400,1,Dial(h323/1131517400@10.11.2.80,10) exten=_1131517401,1,Dial(h323/OPENH323GK@10.11.2.80,10) exten=_2001,1,Dial(h323/2001@10.11.2.80,10) exte...
2010 Apr 20
0
SIP one-way audio
Hi, This problem has been tackled over and over, I know. I'm trying to understand why I'm having trouble with my "simple setup". My setup is like this: <SIP_PROVIDER>---<DSL1>---<LINUX_GATEWAY>---<ASTERISK_VIA_DSL1> I've unloaded the nf_*_sip kernel modules from the LINUX_GATEWAY just in case they could interfere. The DSL1 modem/router is a THOMSON SPEEDTOUCH and I've disabled "SIP ALG". If I place a SIP call then I imm...
2007 May 05
1
SIP registration problem
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2007 Mar 30
4
Speed Dial Application in *
Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar