similar to: chan_sip2 broken with FWD

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..

Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to

Hi, running Asterisk 1.6.2.0 and have started to see in messages: [Jan 14 05:43:43] NOTICE[29231] chan_sip.c: Peer '100' is now Lagged. (4007ms / 3000ms) [Jan 14 05:43:53] NOTICE[29231] chan_sip.c: Peer '100' is now Reachable. (9ms / 3000ms) [Jan 14 05:44:02] NOTICE[29231] chan_sip.c: Peer '100' is now Lagged. (5008ms / 3000ms) [Jan 14 05:44:12] NOTICE[29231] chan_sip.c:

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

Hi, Recently, one of my Asterisk servers stopped connecting calls and required a reboot to "fix it" (did not try to restart or reload). The log showed loads of this message: NOTICE[302] chan_iax2.c: I should never be called! This highly repeated message seems to be preceded by something like: WARNING[10767] channel.c: Exceptionally long voice queue length queuing to

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as reported? Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke: Peer

Hi All, I'm wondering if anyone can share any info on why I frequently get peer timeouts like below, and receive 489 messages from another A*k server on the same LAN. For the peers, we've one L2 switch. ICMP is <1ms. The CPU of the main A*k server is usually < 2%. So I can't see why we'd get such large delays. The phones are all Cisco 7940s (SIP 2xx) The 489 originate

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.

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed authentication. The secret seems correct, so we can't figure out why we're getting failed authentication. But at the same time the device shows as registered: [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 441 [Oct 8 18:15:58] NOTICE[519]:

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

Cam anyone tell me why the condition x[i] == "DISCONECTED" looks like producing an NA instead of TRUE/FALSE I would like to rename "DISCONNECTED" those factors inside the variable "dataset$STATUS.x" that are named "DISCONECTED" thank you > summary(dataset$STATUS.x) ACTIVE DISCONECTED PENDING SUSPENDED TERMINATED 158869 169181

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 =

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br> <br> Using Asterisk 1.4.25.1<br> Using realtime sip_buddies<br> <br> I notice

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

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 get these on a consistant basis for most of the providers I have configured. Some less than others. I even get it from my phone at home to my * box at our data center. What I'm confused about is why it always shows the ping times at right around 2000 ms. That just can't be right. It's always right at 2000 ms. Never less or more by more than 100 or so. May 12 17:42:23

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 have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David