similar to: Params on SIP URI REGISTER/INVITE

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

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..

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

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?

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

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

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

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

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Has anyone successfully implemented concurrent appearance of the same PBX extension on multiple SIP phones? When using Cisco 7960s under call manager, you can have several phones with the same line appearance, but the first user to seize a line makes it inaccessible to other phones. Under SIP operation it seems as though this is not possible, but we don't see group ringing definable for

My first post here, so a brief intro: I'm somewhat new to Asterisk, but have been working with SIP in depth for about 3 years. I studied Asterisk for about a year and have been experimenting with it hands-on for the past month or so. I've done 6 Asterisk installs in wildly different roles/applications, some of them test systems, others in semi-production, so I know a little bit about

On recent releases of the snom200 firmware, the MWI indicator will turn on, but won't turn off when the message has been checked. It works on firmware 2.03o, but not in 2.04g or newer. I filed a bug report with snom, but they're claiming it is an asterisk issue and that it should have been resolved. They suggested that I ask on the list. "Anyway, Asterisk had a bug where it

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

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