similar to: SIP to H.323 Native bridging ...

Ladies and Gentleman, I have setup Asterisk and AMP. They are working with out error. But now I need to get everything going. I have Voicepluse and they give sample iax.conf and extensions.conf files but that does me a little good as I am using AMP. Is there some docs somewhere on how to confgure once I have AMP up and running? I am not a telephony guy and alot of this looks like greek to

Dear Ladies and Gentleman, I am trying to deploy a virtual machine on a KVM host. I created a deployment file (see beneath) and a disk for this VM. As root I executed the following command: virsh create deployment.0 but then I get the following error message: error: Failed to create domain from deployment.0 error: cannot set ownership on /var/lib/one/datastores/0/15/disk.0:

Ladies and gentleman, I'm running Ubuntu Dapper Drake (amd64 version). I managed to set a 32bit Ubuntu dapper chroot on /chroot. It works excellent! on that chroot environment I have installed firefox, in order to run macromedia flash player (i know this is irrelevant to my question). Then I downloaded the latest wine sources from winehq.com website. I compilled and installed it on that

Good evening ladies and gentleman! I'm trying to get Samba 3.0.0 to compile on a HP-UX 11.00 system. Well, "trying to" is the key here... :( I did this do configure the Samba source: ____________________________________________________________ #!/usr/local/bin/bash VERSION=3.0.0 PREFIX=/opt/samba-${VERSION} CC=/usr/local/bin/gcc CPP=/usr/local/bin/cpp

I am just wondering if someone can explain the difference between Packek2Packet Bridging vs. Native Bridging in Asterisk. I'm basically tyring to make sure the media travels end-to-end and I've see both of these bridging types mentioned on the asterisk console. Regards, David

Hi list! The stability of * with fax (or the lack of it) is causing me headaches. To solve it, I was thinking to put a TE205P card in the * box, connect the E1 pri on one port and a channelbank (I was thinking of the Rhino) on the other port in T1 mode. (Has anyone tried this??) The TE205P supports channel bridging which sounded like the ideal (but not cheap) solution when combined with a

Hi, I have the following setup: Asterisk <-> Nat <-> Internet <-> Nat <-> 2 x SIP endpoints With directmedia=no I can make a call between the two SIP endpoints; the RTP stream being passed through the Asterisk box. Obviously, this is sub-optimal. I attempted to enable bridging of the call between the 2 endpoints directly, given that they are on the same non-routeable

Hi, I cannot get two Polycom 600 phones to bridge natively. My sip.conf has canreinvite=yes for both phones. They connect, and I can talk as usual, but sniffing shows the RTP stream is routed through Asterisk. The exact spot where the attempt to natively bridge fails is in rtp.c, line 1281 (CVS from October 8, 2003): f = ast_read(who); if (!f || ((f->frametype == AST_FRAME_DTMF)

Has anyone experienced a dropped call when bridging? I get an "OK, ready to transfer" from both channels, but when asterisk releases the call, it is dropped immediately by the upstream provider. I've tested against another provider and it works fine, and it also works fine across two different providers, including TO and FROM the one that's acting buggy. Here's a

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

Hi all, Is it possible to have native VLAN (802.1q) and a VLAN x at the same time in DOMU''s? This is what I have configured on my DOM0 (xen2): Eth0 (connected by a 801.1q trunk to Cisco 2950 switch) Created vlan100 : vconfig eth0 100 Created bridge : brctl addbr br100 Attach switchport: brctl addif br100 eth0.100 Attach a DOMU : brctl addif br100 vif2.0 I now have bridge br100

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does

I've got several SIP clients with dynamic IP addresses Asterisk has one public and one private IP address SIP clients might connect to Asterisk from either the internet or the private network (192.168.1.255) - they're portable By default, directmedia/canreinvite is enabled and Asterisk sets up direct media connections between clients. In this case clients on the internet can make calls

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

Hi guys, I have a crap load of data to parse and have enjoyed creating a script that takes this data and creates a number of useful graphics for our area. I am unable to figure out one summary though and its all cause I dont fully understand the apply family of functions. Consider the following: #Create data

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

Hi, Sorry,after i install bridging 1.4, even after i setup a new bridge to link between wlan0 & eth 0 but it seem lost after pc restart, i have to reset after restart. am i doing wrong in installation or ... Hope get some advice from u all (expert) ===== Ô¸ÄúÓÀÔ¶ÐÒ¸££¬¿ìÀֺͰ²Ïê Happy & Healthy Always ! ^_^ ! ¸öÈËÍøÒ³ Homepage:http://planet.time.net.my/sunwaycity/ceyong ¸öÈË·ðѧÍøÒ³

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

"she is a great fool for going away, if she liked him."<BR>and the liberty of a manor, it was doubtful to many of those who best knew the easiness of his temper,<BR>and aunt so well, that i am not afraid of requesting it, though i have still something more to ask of the<BR>commerce; but with respect to any other leading characteristic, i do not imagine that much has