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Hi, I am stuck with my extensions.conf and would appreciate a small hint from the ISDN experts. What is the correct way to indicate a busy condition to a calling ISDN zap channel (TE410P) when a local SIP ext. is busy? I have [pstn-in] exten => *****591,1,Dial(SIP/${EXTEN},45,r) exten => *****591,102,Busy and get -- Executing Dial("Zap/1-1",

Hi, I have asterisk connected to PSTN via H.323 gateway via chan_oh323. Incoming calls to SIP extensions work, but SIP message "486 busy here" from a busy extension isn't correctly forwarded to H.323. As a result, a caller from the H.323 side calling a busy SIP extension gets some rings and then an irritating timeout with H.323 message 'no user responding' instead of

hello all, I was trying out the various queueing disciplines and i have problems with this basic one: Simple 3-bands Priority Scheduler --> sch_prio.c how i implemented it: #########START########## tc qdisc add dev eth1 root handle 1: prio tc filter add dev eth1 parent 1:0 prio 5 protocol ip u32 tc filter add dev eth1 parent 1:0 prio 5 protocol ip u32 tc filter add dev eth1 parent 1:0

Hi all, Supposedly my router is itself streaming traffic to clients. Is it possible for it to mark its packets before going through dsmark?? I have gone through the examples in iproute2/examples, i noticed that there''s always an ingress and egress of which these are 2 different dev. Is it possible for before ingress and egress be the same dev?? I tried the script below but

Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

Hi, so far I had our PRI line connected via Digium TE410P and ZAP channel to asterisk which worked perfectly. I now have a second line coming in through a H.323 gateway and chan_oh323. rxfax() still works and receives faxes with G.711 alaw codec, but I always get one empty first page via H.323 - a one-page-fax becomes a two-page-fax with one empty page in front of it. It seems the empty page

Hello asterisk community, I have successfully set up asterisk as a SIP PBX and now would like to connect to the outside world using a Cisco 2600 with VIC-BRI as an ISDN gateway. This works already in the lab, but I have security concerns before conecting the gateway to the internet. I currently don't know exactly what VoIP services the Cisco runs by default besides SIP (H.323, MGCP, ...)