similar to: calling from SIP to a h.323 device with oh323

Hi!, this issue makes me crazy. I read a lot of docs, also * mailling list and I try a lot of things without success. Any help will be appreciated. Here is the info: Hardware: -------------------------------------------- Supermicro Server with motherboard X7DB8, chipset Intel 5000P, Xeon 5050 Digium TE110P Software --------------------------------------------- Asterisk version 1.2.12.1

Hi folks, I'm running asterisk 1.2.10 and I need to use agent dynamic login. I read some doc and follow some tutorials but the agents can't login into the queue. Asterisk ask to me to dial the password agent and after this, it doesn't do nothing ( it doesn't tell login ok or login incorrect..). In the * console if I do show agents, any agent are logged. Any help will be

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

Hi, Anyone is using LG ip phone LIP-100 with Asterisk. I've two of this phones but seems to work only with net2phone, in the product page http://isupport.lge.co.kr/html/ibu_lgic_modelView.jsp?jgrcode=D2_IPTP&modelid=M_IP100C the features are showing SIP and H.323 support. Can be used with my asterisk box? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq.

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi, I'm trying to compile asterisk 1.2.14 on a Debian Sarge amd64 with kernel 2.6.8-12-amd64-k8 make[2]: Entering directory `/usr/src/asterisk-1.2.14/codecs/gsm' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -fomit-frame-pointer -fPIC -c -DNeedFunctionPrototypes=1 -funroll-loops

Hi, I'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

Hi, I using asterisk 1.2.4 on a CentOS with Linux 2.6.9-22.0.2.ELsmp kernel. I've two x100p cards connected, only one card is reconigzed by asterisk. 02:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 02:02.0 Ethernet controller: Davicom Semiconductor, Inc. 21x4x DEC-Tulip compatible 10/100 Ethernet (rev 31) 02:03.0 Communication controller: Tiger Jet

From: "Guillermo Salas M." <gsalas at manta.telconet.net> > http://www.gizmo5.com/opensky Free calls are available up to 5 > minutes. If you need longer calls there's a commercial service you can > purchase. > Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to

Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. <-- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP

Hi, I'm new in this list and my experience on Asterisk quite limited, can anyone help me. This line is shown in the asterisk console and I don't now the meaning -- parse_srv: SRV mapped to host proxysip.sip.somedomain.com, port 5060 Thanks folks -- ---------------------------------------------------------------------------------------- Marc Patino G?mez Dpto. Sistemas Claranet

i use asterisk cvs head ( two days ago) more or less openh323 1.12.2 (oh323 home page) and pwlib 1.5.2 (oh323 home page) asterisk-oh323-0.7.2-pre1 library versions? where download? versions from oh323 readme are not in sourceforge site. but i obtain this error compiling: root@backup:/usr/src/asterisk/cvs/last/asterisk-oh323-0.7.2-pre1# make for x in wrapper asterisk-driver; do make -C $x

Hi, I've oh323 chan installed and working to make calls from SIP to H323 devices. The problem is can no hear sound with the H323 device. I think this is some related with codecs o nat, because the H323 have one public IP from a different subnet from the asterisk box. If I use netmeeting in gateway mode, the call can be completed and I can talk with a SIP device, but in gateway mode I can not

I downloaded and installed the none commercial g729 codec very often now I only disable HT on my systems I think * doesn't like this One of the guys @ digium advised me to turn it of, since they haven't written * to be multi treading any way The codec I download is the http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-pentium4.so It should work fine. Wouldn't know what it

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on