similar to: E1 ingress to SIP egress problem with 183 response

Hi together, i try to compile astrisk-oh323 like described in the Readme - pwlib V1.6.6 (janus) - openh323 V1.13.5 (janus) with make-patch - asterisk V0.9.0 i got the following error gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/usr/src/redhat/BUILD/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c

I am trying to compile the latest channel drivers Can anyone tell me what is wrong ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 gcc -shared -Wl,-soname,liboh323wrap.so -o liboh323wrap.so wrapper_misc.o asteriskaudio.o wrapendpoint.o wrapconnection.o wrapper.o wrapcaps.o make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory

I'm trying to use a simple page function. It starts a MeetMe conference with the devices I've listed, but the devices hang up after 3-5 seconds. After doing some research I found this was a problem, and I needed to remove a (5) from app_page.c Well, my app_page.c didn't have the (5). I did make clean; make install again just in case I had some weird compiled version installed that

Hello everyone, I've had the time to examine some specific serial/parallel forking scenarios with Asterisk lately. Looking at chan_sip it appears that anytime Asterisk wants to tear down a call before it's brought up, it sends a 603 Declined: } else { /* Incoming call, not up */ const char *res;

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

Hi, erverybody The Asterisk is running well in the linux system. Now I would like to add oh323 in Asterisk. I have download pwlib(version is 1.6.6) and openh323(version is 1.13.5). And I sucessfully maked and installed these two packages. But I got the following errors when compling the asterisk-oh323 0.6.3a: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]:

If I have in my dialplan [AgentQ] exten => _XX.,1,Dial(Sip/{$exten},120,g) exten => _XX.,2,NoOP(here we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus:

Hi, I'm chasing a bug in chan_sip.c where Asterisk is removing the rport parameter out of the via headers. Here's my scenario: UA -> Snom NATf -> Snom 4S Proxy -> Asterisk Echo Test Function NATf, the proxy, and Asterisk are all on public IPs. So my question is: In chan_sip.c, copy_via_headers function, I see an if statement testing for "(ast_test_flag(p, SIP_NAT) ==

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

Hello, What''s difference between ingress and egress shaping? When we need to setup ingress shaping(policing) ? Bye

Hi all, Supposedly my router is itself streaming traffic to clients. Is it possible for it to mark its packets before going through dsmark?? I have gone through the examples in iproute2/examples, i noticed that there''s always an ingress and egress of which these are 2 different dev. Is it possible for before ingress and egress be the same dev?? I tried the script below but

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I''ve just discovered that trying to pass traffic from multiple interfaces through the same imq is a Bad Thing(tm). A quick search of "HTB IMQ panic" showed me older posts on this list saying that it''s simply incorrect to pass such traffic through the same imq dev. I''m now back to square one on using the linux

Hi, Can Ingress and Egress can be used in Asterisk, so that Jitter can be calculated...! Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130821/12427233/attachment.htm>

I''m trying to set up customer egress shaping on concentrator that customers connect to using pptp (which uses pppd). This means customers each have a pppXXX interface when they connect. It''s easy to shape their "down" speed this way, and when they disconnect the interface is gone and thus the shaping rules automatically go away as well. However, shaping their upload

i''m trying to limit my egress bandwidth over 2 interfaces (eth1 and eth2) to 2 Mbit my script is something like this: tc=/sbin/ip ipt=/sbin/iptables $tc class add dev imq0 parent 2: classid 2:4 htb rate 1845Kbit quantum 3000 $tc filter add dev imq0 parent 2: protocol ip handle 4 fw classid 2:4 $tc class add dev imq0 parent 2:4 classid 2:40 htb rate 0.5Mbit quantum 3000 prio 5 $tc qdisc

I am successfully running ingress (IMQ) and egress (HTB) shaping on a bridge. Is there any way to combine and share the bandwidth between ingress and egress? Example: I have set up www service for egress at 128 KB and ingress at 256 KB. The shaping on them works fine separately. However, I want to create a single virtual pipe for www traffic and limit both ingress and egress combined to 256 KB.

Hi, Does anybody know how to get the SIP call ID of a "Dial" command? Thanks in advance. Ray -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071008/7f27548e/attachment.htm

Hey all, I've been fighting with this all morning, and I feel like this should be a relatively simple task, but I just can't get it to work. I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip trunk to my carrier. What I'm trying to accomplish is simply adding the asterisk generated SIPCALLID of the leg between asterisk and

I have two interfaces, eth0 and eth1 but i can''t discern the egress traffic than ingress traffic. I need to apply htb qdisc in both directions, and I read that I need the IMQ patch to do this, because in ingress qdisc i can''t apply htb qdisc...but where is the ingress qdisc? affect the traffic that goes from eth0 to eth1 or is for the traffic that goes from eth1 to eth0?