similar to: Cisco & Asterisk

Hey all, I have an as5300 I use for dial in customers, we have 4 PRIs on it. We have a few free channels on it. I'm wondering if I setup SIP on the as5300 I can have asterisk use the free channels for dial out. I'd still have to use my TDM04B for incoming calls, but at least I can expand my outgoing. Anyone done anything like this before? I've never messed with VoIP on Cisco

Hi, I'm trying to detect and or log the duration a a conference (Meetme). I need it in order to do some billing for theses services. Any ideas on how to do it? I googled around but found nothing. Thanks in advance epablo -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup("SIP/test1-a5e1", "") in new

Hi, i'm having another problem I can't work out - make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

I agree, any platform suffers when it is extremely difficult to implement. What we need is an interface that does everything we need and shows what asterisk is capable of, a lot of features will go unused because you might not know the exist unless you hunt them down in the source or conf files. I trained on an Avaya INDeX switch it had a complex console but was laid out in a structured way a

Hi, I've installed dovecot-0.99.10.5 using the rpm from Dag Wieers, but when I start it all I get is nothing (but none of the processes are running). I checked the config file and set it up (I'm trying to use it with mysql support, but it doesn't work with traditional config) Any ideas -- Pablo Endres <epablo at comvoz.com> ComVoz Communications USA: +1 954 343-2085

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

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

Hi people. I've been having some audio problems with some of my cisco 7940 phones using firmware 7.1. The sound gets gargoled up, I can't understand much of what is said (listing on the IP phone). My setup is the following: 7940 - * - Cisco 2621(GW) - T1 I'm using SIP and g729 for all stages of the communication. I was using ulaw for the leg between * and the Gw, but some guys on

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension

http://asterisk.bkw.org/diff/cdr_odbc.diff Now this should fix it the proper way. bkw PS: Thanks for the info klasstek! > -----Original Message----- > From: asterisk-dev-admin@lists.digium.com [mailto:asterisk-dev- > admin@lists.digium.com] On Behalf Of Pablo Endres > Sent: Wednesday, May 19, 2004 3:30 PM > To: asterisk-dev@lists.digium.com > Subject: RE: [Asterisk-Dev]

Hello all, I have a cisco AS5300 which is register with a gatekeeper and a Asterisk server also register with the gatekeeper. PSTN ---->AS5300 ---->Gatekeeper ---->Asterisk I set up a conference room on the Asterisk sever (Room No 1234). I try to call from PSTN to AS5300, The AS5300 will call the Asterisk server through the gatekeeper. I manage to get to the start of the conference

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, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 "Bad Extension" back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-0000004d == Spawn extension (dialin, 065939191, 2) exited non-zero on

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

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, 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

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