The following are the issues resolved in this release:
--- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
--- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)
--- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)
--- Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)
--- push 'outgoing' flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.11.0
Wednesday, April 4, 2012
Asterisk 10.3.0 Now Available
The following are the issues resolved in this release:
--- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
--- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)
--- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)
--- Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)
--- push 'outgoing' flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0
--- Fix potential buffer overrun and memory leak when executing "sip
show peers"
(Closes issue ASTERISK-19231. Reported by Thomas Arimont, Jamuel Starkey)
--- Fix ACK routing for non-2xx responses.
(Closes issue ASTERISK-19389.)
--- Remove possible segfaults from res_odbc by adding locks around
usage of odbc handle
(Closes issue ASTERISK-19011. Reported by Walter Doekes)
--- Fix blind transfer parking issues if the dialed extension is not
recognized as a parking extension.
(Closes issue ASTERISK-19322. Reported by aragon)
--- Copy CDR variables when set during a bridge
(Closes issue ASTERISK-16990.)
--- push 'outgoing' flag from sig_XXX up to chan_dahdi
(Closes issue ASTERISK-19316. Reported by Jeremy Pepper)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.3.0
Sunday, December 11, 2011
PRI for Asterisk
PRI is an abbreviation for Primary Rate Interface and is a telecommunication standard for carrying multiple data or voice DS0 transmissions between 2 physical locations.
All data and voice channels are (ISDN) and operate at 64 kbit/s.
North America and Japan use a T1 of 23 B channels and one D channel. Europe, Australia and most of the rest of the world use a slightly higher capacity E1 of 30 B channels and one D channel.
A B-channel is used to transmit the voice or data and is sometimes also called a user channel, a D channel is used for control messages and signalling.
In some T1 configurations, no D-channels might be used, instead the signalling will be sent inband over the B channels, this is called in-band signalling or bit robbing, resulting in lower transmission rates than the E-carrier system. This resulted in many US ISDN installations only having an effective data rate of 56 kbit/s over a nominal 64 kbit/s channel. See also A&B. This depends on the framing format used.
In the states its common to order a fractional T1, with less than 23 B channels.
1.3. signalling
On both E1 and T1, one timeslot is usually reserved for a D-channel for call setup and call teardown. (called signalling).
- CAS: Channel associated Signalling: with this kind of signalling, a set of bits is used to replicate opening and closing the circuit (as if picking up the telephone receiver and pulsing digits on a rotary phone), or using tone signalling which is passed through on the voice circuits themselves.
- CCS: Common Channel signalling: A more recent kind of signalling, (ISDN signalling and ss7 signalling are a subgroup of CCS signalling.) In this kind of signalling, short messages are sent over the signalling channel, with more information about the call, including caller ID, type of transmission required, etc. etc.
1.4. Framing
- HDB3
- AMI: Aternate Mark Inversion
- B8ZS:
1.5. Timing or clock sources
A PRI connection needs a timing device on one of both ends.
A PRI line can be clocked internally or can be clocked by the telco.
All data and voice channels are (ISDN) and operate at 64 kbit/s.
North America and Japan use a T1 of 23 B channels and one D channel. Europe, Australia and most of the rest of the world use a slightly higher capacity E1 of 30 B channels and one D channel.
A B-channel is used to transmit the voice or data and is sometimes also called a user channel, a D channel is used for control messages and signalling.
In some T1 configurations, no D-channels might be used, instead the signalling will be sent inband over the B channels, this is called in-band signalling or bit robbing, resulting in lower transmission rates than the E-carrier system. This resulted in many US ISDN installations only having an effective data rate of 56 kbit/s over a nominal 64 kbit/s channel. See also A&B. This depends on the framing format used.
In the states its common to order a fractional T1, with less than 23 B channels.
1.3. signalling
On both E1 and T1, one timeslot is usually reserved for a D-channel for call setup and call teardown. (called signalling).
- CAS: Channel associated Signalling: with this kind of signalling, a set of bits is used to replicate opening and closing the circuit (as if picking up the telephone receiver and pulsing digits on a rotary phone), or using tone signalling which is passed through on the voice circuits themselves.
- CCS: Common Channel signalling: A more recent kind of signalling, (ISDN signalling and ss7 signalling are a subgroup of CCS signalling.) In this kind of signalling, short messages are sent over the signalling channel, with more information about the call, including caller ID, type of transmission required, etc. etc.
1.4. Framing
- HDB3
- AMI: Aternate Mark Inversion
- B8ZS:
1.5. Timing or clock sources
A PRI connection needs a timing device on one of both ends.
A PRI line can be clocked internally or can be clocked by the telco.
Sunday, December 4, 2011
USB 2.0 port support heavy telephony traffic
The USB 2.0 interface provides a theoretical speed of 480,000,000 bits per second. A typical uncompressed phone conversation uses about 64,000 bits per second per direction, plus some overhead; in total less than 200,000 bits per second per phone call is used. Thus, the theoretical concurrent number of conversations that the USB 2.0 interface can handle is 480,000,000 divided by 200,000: roughly 2,400 calls for a single USB 2.0 port.
This is a theoretical number, of course, and the typical host processor will not be able to handle this amount of concurrent calls, but it gives a clear answer to the question: "Can the USB port support heavy telephony traffic with multiple Astribanks?"
Connecting multiple Astribank units to a single USB 2.0 port is easy: simply use a USB 2.0 Hub to connect many Astribank units.
This is a theoretical number, of course, and the typical host processor will not be able to handle this amount of concurrent calls, but it gives a clear answer to the question: "Can the USB port support heavy telephony traffic with multiple Astribanks?"
Connecting multiple Astribank units to a single USB 2.0 port is easy: simply use a USB 2.0 Hub to connect many Astribank units.
Xorcom Astribank XPP Technology
Xorcom has developed a revolutionary concept in telephony interfaces for Asterisk systems which we call Astribank XPP Technology. Using USB 2.0 ports to connect to any Asterisk server, Astribank eliminates the requirement for a PCI (E1/T1) card, and even for PCI slots.
The USB 2.0 interface provides a theoretical speed of 480 Mbits per second, which for typical uncompressed phone conversations translates to roughly 2,400 calls per port. The USB architecture employed by Xorcom products affords many advantages, including the ability to build large systems that support hundreds of analog extensions, rapid set-up and easy maintenance.
The USB 2.0 interface provides a theoretical speed of 480 Mbits per second, which for typical uncompressed phone conversations translates to roughly 2,400 calls per port. The USB architecture employed by Xorcom products affords many advantages, including the ability to build large systems that support hundreds of analog extensions, rapid set-up and easy maintenance.
Monday, November 28, 2011
Asterisk ACDs Call Centers
Automatic Call Distributors (ACDs) allow call centers to handle thousands of simultaneous calls, routing them to agents based on caller input, dialed number, load and other factors. ACD systems typically cost tens if not hundreds of thousands of dollars and require specialized training to install and operate. With Asterisk you can build a powerful ACD for the cost of the server hardware and phones.
Asterisk Supported Protocols
Asterisk® supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol (SIP), Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP).
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Using the Inter-Asterisk eXchange (IAX™) Voice over IP protocol Asterisk® merges voice and data traffic seamlessly across disparate networks. The use of Packet Voice allows Asterisk® to send data such as URL information and images in-line with voice traffic, allowing advanced integration of information.
Asterisk® provides a central switching core, with four APIs for modular loading of telephony applications, hardware interfaces, file format handling, and codecs. It allows for transparent switching between all supported interfaces, allowing it to tie together a diverse mixture of telephony systems into a single switching network.
Subscribe to:
Posts (Atom)