similar to: asterisk to use an outbound proxy

Dear all, Does anyone try to install 2 or multiple TE411 card into one server? Can it be done? What about stability? Thanks Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/b65a1d46/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, I want to use Asterisk to use Kamailio as an outbound proxy for routing calls to remote SIP end points, one option could be to use a default peer, but in my case, my outbound proxy can change based on the remote end point, so this option doesn't work. And another problem is that I don't know how to configure Asterisk to prepare the Request-URI based on the remote end point and not

I'm using chan_sip, I experimented with adding a 'Route' header in the originate command and used the Dial command like 'SIP/peer/exten', but problem is that Request-URI isn't populated correctly. I'm using Asterisk 13. Thanks, Nitesh On Wed, Apr 13, 2016 at 10:09 PM, Joshua Colp <jcolp at digium.com> wrote: > Nitesh Bansal wrote: > >> Hello,

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm

Hello !!! I'm trying to setting up my DVG-2032S Voip Gateway to use Asterisk as outbound proxy, that's because I already have this gateway before to begin to play with Asterisk. Every time when I enable the OutBound Proxy option and call from my Ericsson PBX I got the follow message in DVG-2032S System Information: Hook off, and nothing in Asterisk log or in the console. I've

sorry if this has been asked before. is it possible to configure asterisk to use an outbound sip proxy? thanks __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

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

OK, it's been a while since I drank from the pool of wisdom hear on the list. After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

Hi There, We have single PRI with multiple DID numbers and its working fine in receiving call. And if you make outbound call it will send main-line CallerID (company name). Now we want individual caller id for per extensions on outbound calls. like if i call someone he will get my extension as callerid ( 617-838-XXXX) XXXX is my sip extension something like this so next time i direct get call

I'm running a pretty successful Asterisk system and recently moved our PRI to a T100P board. The PRI was previously connected to a Cisco 2600 that was serving as a voice gateway. We are having a frequent problem with inbound and outbound calls being disconnected shortly after they are answered since moving the PRI directly to the Asterisk box. Most calls work fine, but approx 3 out 10 are

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

Hi All, Simple scenario: 7940 SIP><NAT Router><INTERNET><Asterisk SIP B2BUA w/Public IP Inbound/outbound calls work fine 2 way audio, features ok, no issues that I can tell so far. 7940 SIP Using Outbound SIP Proxy><NAT Router><INTERNET><Asterisk SIP w/Public IP Phone registers, call in/out SIP Signaling traversing the proxy ok no audio on phone, SDP

Is there a way to log outbound connections to a specific port (80)? CentOS 4.6. iptables? Thanks Tony Schreiner Boston College

In the latest release of chan_sip2 I've added support for SIP Outbound Proxy. I've seen a lot of requests for that lately, so if you can test this and confirm wheather it works for you or not, I'll be grateful. If I get positive reports, we'll try to add this to chan_sip in CVS. It works like this: * Configure outboundproxy in the general section of sip.conf outboundproxy =

I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account

Hi i need to register with my pstn server but this use outbound proxy, i have been reading the wiki but not figure how configure de outbound proxy only say the sentences is this ... register => user:pass:fromuser@server:port/# but where specific the outbound proxy ? my isp provider WWTelco don't work if i not specific this parameter. Thanks a lot, and sory for my english George