similar to: capacity testing

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

Hi folks, The obligatory newbie disclaimer: "Hi, I'm new to Asterisk and I have a couple questions..." OK, now that that's over with: I've just started working for a small CLEC, and I'm trying to sell * to my boss as a replacement for some expensive/inflexible/closed-source software he's been using to provide residential dialtone with for a couple years now.

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

Guys. I need some advice on some h323 issues. I need to test connectivity from Asterisk to a Cisco AS5300 that has PSTN lines and to cisco 3600 voip routers. H323 needs to be used here but I was wondering if anybody has linked Asterisk to these Cisco routers before? Thank you for any pointers.

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many "answers." I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the

Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement ("number is not in service" etc.) when I'm calling a disconnected number through

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

the problem below keeps recarrying even after i have cleared this error when i run asterisk -vvv or -c the error occurs again please help ..Warning, flexible rate not heavily tested! .................WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/local/lib/libh323_linux_x86_r.so.1: undefined symbol: _ZN13PASN_Sequence17PreambleDecodeXERER11PXER_Stream WARNING[1024]: File