similar to: Re: SIP re-invite

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

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..

Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. Any insight would be appreciated. Thanks

I have successfully installed and configured the asterisk, the incoming and the outgoing calls are working fine, its a tremendous solution :) Now i want to enable trunking between two asterisk boxes, in the iax.conf i have put: [karachi] ... ... ... trunk=yes ... ... ... everything seems to work fine but when i load asterisk it says: -------------- Feb 17 10:59:14 WARNING[18726]:

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

After browsing through bugs.digium.com I saw no mention of any work to get chan_sip or chan_sip2 to listen on tcp, as well as udp. Just curious is any-e-one working on such a patch at the moment?

I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] => (Inter Asterisk eXchange

I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]:

Good evening from the CVS of the 2005/05/14 it's impossible to build asterisk* on a redhat 7.3 i get this at compile time chan_sip.c: In function `build_user': chan_sip.c:10007: parse error before `struct' chan_sip.c:10029: `userflags' undeclared (first use in this function) chan_sip.c:10029: (Each undeclared identifier is reported only once chan_sip.c:10029: for each

I'm trying to get call parking working with the lighted buttons on the SNOM 200. I have set the 5 buttons to "Park Orbit", for extensions 700-704. Pressing the first button (x700) does park the call. However, the remaining buttons (x701-704) don't allow me to pick up parked calls, or show parking status via the LEDs. I can only pick up parked calling by manually dialing the

I've a problem connecting uniVoice (http://voice.uni.it) from asterisk. Using my account data I can place a call smoothly using xlite or my budgetone phone directly, but I'm not able to use uniVoice as a peer from asterisk. Registration seems to work correctly, but when I try do dial, the sip authentication fails every time. Their tech people told me that they are unable to make

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

Hi all! So far I've always used plaintext passwords for SIP, but now I've decided to use MD5 encryption. For each client I edited its section as follows, then: auth=md5 md5secret=hashed_passwd ;secret=plaintext_passwd where hashed_passwd is the output of echo -n "user:realm:plaintext_passwd" | md5sum When the first SIP clients registers with Asterisk after a "sip

Hi all, I try since few days to register to FWD with chan_sip2 and always been disconected: peer TOO LAGGED and then peer is now REACHABLE! and so on. Now I restart asterisk with chan_sip and get it work. So for me it's chan_sip2 which is broken,*only with FWD* (at least for me) as I have others SIP providers and it's working fine with them. I use a CVS version of asterisk from

We're doing some SIP development and have a question on "additional parameters" supplied to the register (in this case maddr= and the non-standard clport= in our example below). What we're experiencing is the INVITE doesn't included these parameters and they get dropped when the INVITE is sent to the 10.1.1.97 address. Ideas? Supported? SIP Bug? REGISTER

I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xxxxxx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't it send out the same "a=rtpmap:103 telephone-event/8000" to the other side of the connection? and not something like "a=rtpmap:101 telephone-event/8000"? Thanks

Hi I''ve spec''d a class and they pass. Now I''d like to assure that any subclass of this class also passes the same specs. Any suggestions for a clever way to handle this? I''d prefer to keep the existing specs as is (eg instead of moving everything into shared behaviors, or doing something to all the ''describe'' lines) thanks linoj

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

*** Astricon 2004: Over 250 Asterisk professionals! Astricon, the first Asterisk user's and developer's conference is a success and we now have over 250 people registred. Thank you for all your support of this event and please have patience with us as we're trying to handle all details with payments, hotel reservations and other issues. I also wish to thank all our partners that