similar to: Asteriks as SIP<>H323 Proxy?

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

I am newest to this group and would appreciate your help! Is it possible to use quicknet phone jack with asteriks@home ver 0.6? Little has been mentioned about use of quicknet products' adaptability with asteriks@home I do have a couple of old jacks to startup right away. Your guide is most welcome. Thanks, Mike __________________________________ Celebrate Yahoo!'s 10th

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on "attended transfers". What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

How can I monitor asterisk if all lines are registered etc? I have an asterisk on a remote location and sometime they reporting problems that phone is not ringing, they can not dial out etc. Usually I just restart asterisk and it solves the problem. Is there an application that will email me if case any line looses registration with with asterisk? Or any better solution! -- Joseph

Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus

Hi There, we are developing a dialer application using the java lib to interface with the asterisk manager protocol. It works fine so far. The only problem we have is that if we use the "originate" command the user is required to pick up the fone _bevore_ asterisk will originate the call to the desired destination. What we would like to do is to place the call, check if the other end

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Mandi! Rowland penny via samba In chel di` si favelave... > yes, Provided you use the right attribute to search on ;-) Ah! ;-) Just i'm here, i test three condition in account flags, eg: UAC=$(ldbsearch ${LDB_OPTS} -b "${BASEDN}" "(&(objectClass=user)(sAMAccountName=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-)

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

I have a Steam version of DAO + Awakening and have had no troubles with it up until now when I've bought some DLCs from official bioware store. For any DLC I bought I'm getting "Unable to load area" error right after character creation or importing (or, in case of "Leliana's song" --- right after "Play" is pressed in "Other campaigns" menu). Its

I hopefully cleared all SAMBA files and set up a fresh ADC using: samba-tool domain provision --use-rfc2307 --domain=UAC --realm=UAC.MGR --server-role=dc --dns-backend=SAMBA_INTERNAL --targetdir=/srv/files --adminpass="secret" --option="dns forwarder=172.16.6.11" The provisioning seemed okay, i.e. nothing hints at any errors and I see a DOMAIN SID as the final entry as

I need to do some testing, but before to hit by head on a known wall, i ask here. My AD domain get used (via PAM/Winbind) to give access to some other dervice, most notably here dovecot. When password expire (or users change it) the MUA try the old password some times, then ask for a new password; users cleraly get scared, press randomly 'OK' or 'Cancel', but if they press 2-3

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Hi, I'm trying to use the function "error.bars", but the program don't find it, and I dont't found any package with this function. Is there some another functin to draw barplots with error bars? Sueli Rodrigues Eng. Agr?noma - UNESP Mestranda - USP/ESALQ PPG-Solos e Nutri??o de Plantas Fones (19)93442981 (19)33719762

Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much.

Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for

Hi I am trying to setup asterisk to do a mass deploy of some snom phones. I can't find where i configure asteriks to listen to the multicast address, nor where to set the notify reply. I was hoping to not have to use dhcp options alex -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc:

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at