similar to: SIP Outbound Proxy Support

Hello, new to asterisk and trying to set it up to 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

It looks like chan_sip2 may solve my problem with outboundproxy support. However, I am having problems getting the solution installed. From what I understand these are the tasks... Add chan_sip2 to the channels/Makefile * Rename the file downloaded to chan_sip2.c * make / make install * Change your modules.conf Add "noload=chan_sip.so" if you want to run chan_sip2 * Restart

I managed to upgrade the phone to 8.2.2SR1 after renaming jar70sip.8-2-2ES1.sbn to Jar70sip.8-2-2ES1.sbn but the phone would continually say "Registering" and the red X next to the phone icon. The phone would eventually time out and couldn't make incoming or outgoing calls. Then I disabled registering with the proxy with the following line in the config:

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, 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

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a community, spending extra time on finding bugs, solving issues, improving

Hi, I've been trying to get asterisk to use an outbound sip proxy. Putting the outboundproxyhost directive in the [general] section of sip.conf, but it doesn't seem to work. My expectation would be that by setting outboundproxy and outboundproxyport in that location, then all dial commans (or at least dial commands of the form Dial(SIP/asdf@asdf.com) or Dial(SIP/asdf@99.99.99.99)) would

Looks like outboundproxyport doesn't support in 1.4.4 If you set the port, then it conflit with the one in "To URI" with host. I saw the code for outboundproxyport from the source, but is it a bug? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070625/7aad927d/attachment.htm

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

Hello List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with 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

Hi, I'm going to forward SIP request to special outbound SIP proxy with none SIP port. I have this in my sip.conf [sip_proxy-out] type=peer ; we only want to call out, not be called username=408 host=192.168.0.95 outboundproxy=192.168.0.74 port=9097 I want a To: 408 at 192.168.0.95 by proxy 192.168.0.74:9097 but it turns out the "To" also has the port To: 408 at

Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!

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Trouble getting chan_sip2 to compile below is what I have done -download acl.c.patch,acl.h.patch,chan2s_sip.c to /root/software cp /root/software/chan_sip2s.c /usr/src/asterisk/channels cd /usr/src/asterisk/ patch -p0 acl.c /root/software/acl.c.patch cd /usr/src/asterisk/include/asterisk patch -p0 acl.h /root/software/acl.h.patch - added the follow to /usr/src/asterisk/channels/Makefile

Hi, Does anyone have any detailed install instructions for setting up chan_sip2.. I patched acl.c but could not see an acl.h file to apply the patch.. I copied the chan_sip2.c file into the channels directory.. I am not sure what I need to do exaclty in the Makefile to get chan_sip2 to build.. Any help and anything to be careful of in chan_sip2 would be usefull.. Thanks, Later..

I am tired of nat tricks, and would really like to run ser on a system that straddles the internal and external network, and send all outbound sip traffic to it (it would also rtp proxy). This would also give the huge benefit of actually being able to implement SIP reinvites some of the time, even though the * server is behind a nat. I know there's no outbound proxy support in chan_sip

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com