similar to: Problems with calls in asterisk.

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070126/a49e3bdb/attachment.htm

I am working with implementing Asterisk between four different AS5400's located in multiple sites with different PSTN gateways. I can get two of them to work without a problem, but I am getting the following on the others when I make a SIP call to the other two sites. Got SIP response 500 "Internal Server Error" back from 10.1.3.28 SIP/alma-1b77 is circuit-busy Everyone is

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

I am currently using Asterisk with G.711 codecs and in-band DTMF for several Cisco 7960's and an Audiocodes GW. When allowing out-of-band DTMF, I could use voicemail menus and anything else on Asterisk that required DTMF but I could not get the DTMF relayed out of the GW. Has anyone verified that this works between 2 SIP devices? If so, I would be interested in your settings. Also, I would

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i have a working Microsoft ISA firewall with buildin H.323 Gatekeeper.... So Far, i got registerd the asterisk on the M$ Gatekeeper... here is the h.323 configuration: ; Open H.323 driver configuration ; [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs dtmfmode=rfc2833 gatekeeper =

Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson.

I have built Asterisk on SuSe Linux 7.3 on an Ultra 2 Sparc WorkStation. I am listing the modification I had to do for the benefit of anybody else who wants to use Asterisk This workstation is equipped with one 400 MHz RISC UltraSparc II CPU, 256 MB RAM, Two 9 GB 10,000 RPM UltraSCSI Disks. I have a gatekeeper running on this machine, I had to do the following modification to build * on Sparc:

Hello, I am trying a setup that is the following: SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) ----> PSTN Any calls from H.323 GW through GK goes to PSTN, no problem. SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem. SIP Phone to PSTN, rings normally, on the PSTN, then connects when the PSTN phone picks up, no audio on both directions. PSTN GW support

Hi, I''m new to XEN and have the following Problem: If I shutdown my domU the network of the dom0 leaks. A ping from another server to my xen dom0 shows a Packetloss > 80%. Then if I restart the domU after some seconds the ping show a loss from 0%. I retryed that 3 times but it is really hard to go to the dom0 via ssh and recreate the domU if the packetloss is that high. I used google

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Hi all, I'm trying to install * on Mandrake 9.2/P4, but under asterisk - make clean;make install there is the following error: ---------------------------- [root@net asterisk]# make clean for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving

Hello, Thank you for the job well-done. I installed the chan_h323 of the asterisk-1.2.7.1 and with lib pwlib-v1_10_0-src-tar.gz and openh323-v1_18_0-src-tar.gz and I used licensed g729 from digium. However, I am having a very funny behavour. 1. If I send a call on its ringing at the called side but the caller didn't get the ringing tone. 2. if the called picks up the phone, I am

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

Hi, compiler-rt builtins currently doesn't build on MSVC 2019, I the problem is that compiler-rt\lib\builtins\int_math.h includes the header ymath.h. according to eg. https://docs.microsoft.com/en-us/cpp/c-runtime-library/reference/finite-finitef?view=vs-2019 the header to include is float.h also the ymath.h file contains the comment /* ymath.h internal header */ so probably shall not be

Send these questions to Asterisk-Users mailing list. h323.conf ################################################## ; ; Configuration file of OpenH323 channel driver ; [general] listenAddress=W.X.Y.Z ; local ip listenPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=yes jitterMin=20 jitterMax=100 ipTos=none outboundMax=100

Hi! everybody: We used opus-codec for a VOIP gateway. The GW is running at a UBUNTU server. The opus stream is transcoded to G711 pcmu stream.So there are many opus codecs running simultaneously. We noticed that if there more than 5 streams in. the voice then has notisable glitchs.More streams in, worse voice got. Then we write test code for opus-codec which encode a .pcm file simultaneously.