search for: chan_sip2s

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

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 chan_sip2.so: chan_sip2.o cd /usr/src/asterisk make make install...

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..

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

Hello, I've tried to compile the new sip channel, sip_chan2.c but I am not succesfull. When I make * I get error messages, some of them also considering syntax error in the code. Does anyone use this channel? Wuld you please give me some advices how to compile it? Or do you have the source code that works??? Thanks for answers. Tomas

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?

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

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

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

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

*** 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

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

I'm having trouble connecting a asterisk server to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm

Welcome to a new issue of Asterisk Summer News! The holiday season is coming to an end here in Sweden, people are getting back to work and the kids will start going to school next week. Life is slowly adopting to normal and I have to start dressing more towards a businessman than a beach bum. Guess I have to start going to the gym again as well. Anyway, back to the topic. Asterisk and VoIP.

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 get this warning from Asterisk and I want to assess whether it is important, and if so, if I should complain to the telephone manifacturer or start up my programmer's editor: chan_sip.c:5152 handle_response: Host '172.31.1.7' does not implement 'NOTIFY' What does this mean and am I missing some important feature? Groeten, Joost Kraaijeveld Askesis B.V.

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

Is there any reason why we wouldn't want to add a setting for each REGISTER => line in sip.conf that allows us to set refresh time on a per-proxy basis? If not I thought I would try to add that functionality to chan_sip. Please tell me if there's some reason why we wouldn't want to do this, (e.g. it's coming in a future CVS or in chan_sip2 or it will horribly break the code

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 =