search for: outboundproxy

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- A...

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of the "outboundproxy=" IP as it should, verified with tcpdump. I've done a search of the mailing list and didn't turn up anything r...

...93512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 p...

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69....

Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls sho...

...o work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not sure if it's an issue but my voip provider uses port 80 for sip instead of 5060. In asterisk, I can make outgoing calls through my voip provider to pstn lines, audio works both ways. But calling in from my land line to the asterisk box via vbuzzer, I get n...

Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general section this: outboundproxy=192.0.2.1 Works OK , but now Asterisk sends all SIP messages...

...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 you give me advice what I should set, when I...

...ith Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the SBC to the voice switch defined by URI. [username] type=peer secret=password host=sip.example.com outboundproxy=sbc.example.com context=from-ISP-X From the Dialplan that string is dialed: Dial(SIP/username/${EXTEN}) This works fine, asterisk sends the call to the outboundproxy defined in the sip.conf section of [username]. Before adding outboundproxy setting, incomming calls were matched because they ori...

...sm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register => SIP-ID:SIP-Password at sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_conf...

...t pointless to show you my settings. The one I've been playing around with most recently is: [voipexten] auth=00575000010XXXX:00575000010XXXX at las-obproxy.voipzone.us username=00575000010XXXX secret=00575000010XXXX fromdomain=directnationalloan.com type=peer qualify=yes insecure=port,invite outboundproxy=las-obproxy.voipzone.us But of corse that doesn't work. Maybe someone here has an idea. -- /Nick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/77246474/attachment.htm

...rot.SIP.lcs="0" voIpProt.SIP.sendCompactHdrs="0" voIpProt.SIP.WM50="0" voIpProt.SIP.keepalive.sessionTimers="0" voIpProt.SIP.requestURI.E164.addGlobalPrefix=""> <outboundProxy voIpProt.SIP.outboundProxy.address="10.0.20.0" voIpProt.SIP.outboundProxy.port="5060" /> <alertInfo voIpProt.SIP.alertInfo.1.value="" voIpProt.SIP.alertInfo.1.class=""...

..." reg.1.type="private" reg.1.lcs="" reg.1.thirdPartyName="" reg.1.auth.userId="ip4000_1" reg.1.auth.password="password" == Polycom company-wide config on TFTP server == server voIpProt.server.1.address="x.x.x.55"/> <SIP> <outboundProxy voIpProt.SIP.outboundProxy.address="x.x.x.55"/> </SIP> I've tried using x.x.x.55 as both the "proxy" value only, the "server" value only, and (in the given example) both. I also added the following to sip.conf, to no avail: deny=0.0.0.0/0.0.0.0 permit=x...

...; 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222...

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

...; register => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=luca_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493511111111 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [pbxfax] > type=peer > defaultuser=00493512222222 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=fa...

...493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222...

...ahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten => 22,1,Dial(SIP/114186939930 at pri1.omnity.net,30,D(132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs...

...30"/> <SIP voIpProt.SIP.useRFC2543hold="1" voIpProt.SIP.lcs="0" voIpProt.SIP.sendCompactHdrs="0" voIpProt.SIP.WM50="0" voIpProt.SIP.keepalive.sessionTimers="0" voIpProt.SIP.requestURI.E164.addGlobalPrefix=""> <outboundProxy voIpProt.SIP.outboundProxy.address="" voIpProt.SIP.outboundProxy.port="5060"/> <alertInfo voIpProt.SIP.alertInfo.1.value="AA" voIpProt.SIP.alertInfo.1.class="3 "/> <alertInfo voIpProt.SIP.alertInfo.2.value="RA" voIpProt.SIP.alertInfo.2...

..."/> <SIP voIpProt.SIP.useRFC2543hold="1" voIpProt.SIP.lcs="0" voIpProt.SIP.sendCompactHdrs="0" voIpProt.SIP.WM50="0" voIpProt.SIP.keepalive.sessionTimers="0" voIpProt.SIP.requestURI.E164.addGlobalPrefix=""> <outboundProxy voIpProt.SIP.outboundProxy.address="" voIpProt.SIP.outboundProxy.port="5060"/> <alertInfo voIpProt.SIP.alertInfo.1.value="AA" voIpProt.SIP.alertInfo.1.class="3"/> <alertInfo voIpProt.SIP.alertInfo.2.value="RA" voIpProt....