search for: techhat

...kefile chan_sip2.so: chan_sip2.o cd /usr/src/asterisk make make install I assume that problem is with what did or didn't add to the Makefile Thank for any help ----- Original Message ----- From: "Olle E. Johansson" <oej@edvina.net> To: "Glenn Dalgliesh" <asterisk@techhat.com> Sent: Tuesday, April 20, 2004 1:29 PM Subject: SIP re-invite > Could you please test this with my chan_sip2. I have a hunch it will work with > that channel. > > /Olle >

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 am try to come up with a way to dial multiple ext and require one or more of the extension to require input before actually bridging the calls. Example: A call from PSTN into * a match is made in extensions.conf and * then dials a local(fxs) ext 1 and a Cell Phone If ext 1 picks up * bridges the call if Cell picks up require digit to in order to bridge the call if neither occurs and timeout is

I have been doing some testing and have found issue with certain devices and negotiating codecs in doing this I have noticed something that seems peculiar to me. It seems that including allow=all yields different results than having no disallow or allows in the sip.conf. Could someone please explain why that is true? Thanks -------------- next part -------------- An HTML attachment was

I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is forwarded to cell phone but If cell is out of range, turned off,

Sorry, If this is a dual post, was having trouble with email. I am looking for a way to Forward to a external or internal number and require a digit(s) in order to complete forward. Example: PSTN1 Calls * dials PSTN2 if PSTN2 presses proper digits bridge the PSTN1 and PSTN2 if no response return to a context Reasons: 2 actually 1. call is

When a call is place to xxx9931211 from the pstn the call proceeds normally until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of call being sent with INVITE sip:xxx9931211@proxy.yyyyy.net SIP/2.0. It gets sent with INVITE sip:xxx9931211@yyy.33.165.201:5060 SIP/2.0 and this seems to cause SNOM proxy to return the packet without a Record-Route entry and then asterisk starts

Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel drivers but that is about it. Just looking for some consensus to if any problems exist with it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/56d03b72/attachment.htm

After setting the pridialplan=unknown I seeing the Called Number TON change to Unknown Number Type but not the Calling Number TON. Should both be following this parameter or not. If not is their another option to change the Calling Number TON? > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) > Ext: 1 Progress

Just woundering if the intentend functionality of leastrecent and fewestcalls it to continually dial only the first chosen ext. in the queue. In other words if a memeber is logged into the queue but doesn't answer the call the call never moves on in my configuration from that ext. This could be really bad!!!! Thanks [support] announce-frequency=45 strategy=leastrecent music=default

Could some please confirm that this behavior is incorrect. I am seeing issues where it appears that asterisk is not following the Record-route on it's reply messages. Please let me know if you require any other information. Thanks Example: xxx.yyy.154.243(PSTN-GW) <--sip--> xxx.yyy.77.23(Asterisk) <--sip--> xxx.yyy.91.74(SNOM or SER proxy) <--sip---->

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

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarat ions -g -Iinclude

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

I am seeing an issue with getting certain sip devices to be recognized as defined SIP clients host= in the sip.conf and the only deference that I can find btw sources that work and don't work is that devices that send packets with an Initial Via header of themselves appears to work and pick the context correctly but those that don't have the Via just get dropped in the context of the

I have an IconnectHere account with a Inbound number and have setup the sip.conf to register and am recieving the call but When I answer the call it disconnect. I have tried sending the call to from * to a Softphone, Pingtel, and FXS port and all result the same. As soon as I accept the call it disconnects. I believe it may be some type of codec issue but I am not very familiar with that layer.