asterisk-users (page 38 of 413)
- fax pass-through, (continued)
- SIP Register,
Tomislav Parčina
- audio cuts out,
trixter aka Bret McDanel
- voicemail recording format,
Vincent Régnard
- uniden uip200 loosing registeration,
Jerry Geis
- Developing a call centre app. Communication with asterisk?,
Arne Morten Johansen
- Telmex PRI line configuration problem,
Oscar Carriles
- Asterisk and MOH for Queues,
Waldo Rubinstein
- SIP Header VIA when behind NAT,
Jean-Marc Salsa
- about g729 license,
Dov Bigio
- Help Asterisk with Phoneserve,
Carlos Rojas
- Lucent Avaya Partner ACS T1 module,
C F
- Can Asterisk send RTP to a specific port number?,
Jimmy
- consult about Digium Card,
Luz Lopez
- Planet VoIP Phones,
Andrew Kirch
- Solution for 1 time blast of 200, 000 recorded calls,
Ron Senykoff
- ChanIsAvail,
Jayson Navitsky
- consult about Digium Card,
Dean Collins
- Guidance need for trunking using SIP,
John Joseph
- Dial command to connect two channels and bypassasterisk server,
Wai Wu
- SPA-941/2 Monitoring,
Josh Dady
- Nat, SIP, Realtime problem,
Hall, Eric M.
- Use one sip account for multiple sipura,
Reli Loin
- Rough Two Days,
Sean Cook
- Solution for 1 time blast of 200, 000 recorded calls,
Dean Collins
- Podget or Similar,
Bob McDowell
- Bristuff-0.3.0-PRE-1l and TDM400 with fxo ports,
Allan Gee
- [help] warning 4246,
fabrizio
- echo problem,
asterisk183
- Instant Messaging: with SIP or XMPP,
roswel ajf
- Nat, SIP, Realtime problem,
Hall, Eric M.
- Dial command to connect two channelsand bypassasterisk server,
Wai Wu
- Traffic prioritization and 'class of service' for SIP,
Philip Edelbrock