I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time? - At least 4 PRI ports - Act as Network and User - SIP Gateway Thanks! -Jonathan
Jonathan Thurman wrote:
I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time?
Cisco AS5300 or other Cisco chassis with voice feature cards is unmitigatedly solid, stable and good for this purpose. Fully loaded, it can do 4 PRIs - network or user side. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
The 4 PRI makes it a little harder than the initial statement of 2. Something like an AS5400 or an AudioCodes GW might be your only decent options. If 2 were enough, you can look at the AdTran TA900e Series or Edgemarc 4500T4's. Both of these have 4 T1 ports max, but I don't think either of them support 4 PRI's at once, regardless of role. -Scott -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jonathan Thurman Sent: Thursday, October 29, 2009 3:52 PM To: VoiceOps Subject: [VoiceOps] PRI Termination device I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time? - At least 4 PRI ports - Act as Network and User - SIP Gateway Thanks! -Jonathan _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
On Thu, Oct 29, 2009 at 1:06 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
Cisco AS5300 or other Cisco chassis with voice feature cards is unmitigatedly solid, stable and good for this purpose.
We currently have some 2800 series devices we are looking to replace. Besides capacity, how to the AS5300 devices compare? I must say that while the Cisco devices do work, I don't prefer them. On Thu, Oct 29, 2009 at 2:23 PM, Scott Berkman <scott at sberkman.net> wrote:
The 4 PRI makes it a little harder than the initial statement of 2. Something like an AS5400 or an AudioCodes GW might be your only decent options.
Right now we need 2 User and 1 Network PRI. Planning for limited future expansion is what went to 4. I would consider splitting onto multiple devices for hardware redundancy though... -Jonathan
Jonathan Thurman wrote:
On Thu, Oct 29, 2009 at 1:06 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
Cisco AS5300 or other Cisco chassis with voice feature cards is unmitigatedly solid, stable and good for this purpose.
We currently have some 2800 series devices we are looking to replace. Besides capacity, how to the AS5300 devices compare? I must say that while the Cisco devices do work, I don't prefer them.
I must say, they really are among the most stable and featureful I've run into. There are lots of other options - and I can hardly claim to have sampled a vast diversity - but the common thread I've run into with other equipment is bugginess, SIP interoperability and DSP issues. I don't know the grounds for your objection to them, but I can tell you they are quite ubiquitous inside carrier environments for a reason. It's hard to say how the 5300 compares to a router outfitted with voice cards; it depends on many things, including the processing power needed for the intended application. It's definitely more powerful CPU-wise than a 2800, though. Can anyone else on the list speak more to the precise feature set differences?
Right now we need 2 User and 1 Network PRI. Planning for limited future expansion is what went to 4. I would consider splitting onto multiple devices for hardware redundancy though...
I'd get a 5300. -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
Might be worth adding a Dialogic IMG 1004 as well to you list of kit to look at. The new 29xx Cisco routers have been released with higher density PVDM3 modules. The AS5300/5350/5400/5350XM etc type devices have a pretty long history. I believe they were designed as more of an access product, rather than the 2800/2800 integrated services routers. I'm not sure if the 2800's do things like resource-pool discrimination and 56k dial termination, although some of these things are not as important as they once were. The AccessServer platforms are the workhorses of dial solutions, dial-and-voice, and with MGCP various SS7 interconnect solutions... On 30/10/2009, at 6:22 AM, Jonathan Thurman wrote:
I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time?
- At least 4 PRI ports - Act as Network and User - SIP Gateway
Thanks!
-Jonathan _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Look at the audio codes boxes. Good stuff. RJ Sent from my iPhone On Oct 30, 2009, at 1:30, Peter Childs <pchilds at internode.com.au> wrote:
Might be worth adding a Dialogic IMG 1004 as well to you list of kit to look at.
The new 29xx Cisco routers have been released with higher density PVDM3 modules.
The AS5300/5350/5400/5350XM etc type devices have a pretty long history. I believe they were designed as more of an access product, rather than the 2800/2800 integrated services routers. I'm not sure if the 2800's do things like resource-pool discrimination and 56k dial termination, although some of these things are not as important as they once were. The AccessServer platforms are the workhorses of dial solutions, dial-and-voice, and with MGCP various SS7 interconnect solutions...
On 30/10/2009, at 6:22 AM, Jonathan Thurman wrote:
I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time?
- At least 4 PRI ports - Act as Network and User - SIP Gateway
Thanks!
-Jonathan _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I'll go ahead and say it: You could look at using a 4-port Digium or 4 port Sangoma T1 card with an Asterisk or FreePBX setup. Jonathan Thurman wrote:
I am looking for suggestions for a device to terminate some PRIs. We are looking to bring in two PRIs from a vendor, and want to provide another PRI out to an on-premise device and attach to our VoIP solution over SIP. Are there any devices that accomplish all of these at the same time?
- At least 4 PRI ports - Act as Network and User - SIP Gateway
Thanks!
-Jonathan _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
On 10/30/09 9:18 AM, Lee Riemer wrote:
I'll go ahead and say it: You could look at using a 4-port Digium or 4 port Sangoma T1 card with an Asterisk or FreePBX setup. This is what my company uses. Some would say this isn't "carrier class" and that's another debate. But it does work, isn't expensive, and the products are readily available.
Carlos Alvarez wrote:
On 10/30/09 9:18 AM, Lee Riemer wrote:
I'll go ahead and say it: You could look at using a 4-port Digium or 4 port Sangoma T1 card with an Asterisk or FreePBX setup. This is what my company uses. Some would say this isn't "carrier class" and that's another debate. But it does work, isn't expensive, and the products are readily available.
But considering you can get something that *is* carrier-grade off the secondary market for about the same price... -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
On 10/30/09 10:19 AM, Alex Balashov wrote:
But considering you can get something that *is* carrier-grade off the secondary market for about the same price...
I don't disagree, and there's much to be said about an appliance versus build your own. I like appliances. We use the Asterisk solution because Asterisk is already the basis of our network, and we have a large amount of Digium hardware. If I didn't have Asterisk/Digium already, I would probably lean towards an appliance.
Carlos Alvarez wrote:
I don't disagree, and there's much to be said about an appliance versus build your own. I like appliances. We use the Asterisk solution because Asterisk is already the basis of our network, and we have a large amount of Digium hardware. If I didn't have Asterisk/Digium already, I would probably lean towards an appliance.
True. In your case, if some PC hardware fails, you've got bigger problems anyway... -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
An SSD and passive heatsink will go along way. You could find some redundant PSUs as well. One problem with 3rd market equipment is getting one with the correct software version to support what you need. We noticed our AS5350 couldn't handle CNAM with the IOS it came with. Alex Balashov wrote:
Carlos Alvarez wrote:
I don't disagree, and there's much to be said about an appliance versus build your own. I like appliances. We use the Asterisk solution because Asterisk is already the basis of our network, and we have a large amount of Digium hardware. If I didn't have Asterisk/Digium already, I would probably lean towards an appliance.
True. In your case, if some PC hardware fails, you've got bigger problems anyway...
On Fri, Oct 30, 2009 at 10:23 AM, Carlos Alvarez <carlos at televolve.com> wrote:
On 10/30/09 10:19 AM, Alex Balashov wrote:
But considering you can get something that *is* carrier-grade off the secondary market for about the same price...
I don't disagree, and there's much to be said about an appliance versus build your own. ?I like appliances. ?We use the Asterisk solution because Asterisk is already the basis of our network, and we have a large amount of Digium hardware. ?If I didn't have Asterisk/Digium already, I would probably lean towards an appliance.
The backend in this case is Asterisk, so a Digium card is not out of the question. Don't be afraid to suggest OSS, it is more pervasive than we all think. Currently we are using multiple Cisco 2800 series devices which work. My hesitance toward Cisco is more about how we are treated than the hardware. Although I don't like some behaviour of the SIP stack on the 2800s. Having an appliance does offer some ease of configuration, although somewhat limits flexibility. On the other hand it allows me to isolate roles, and not have all my eggs in one basket. In the end, we are trying to fix an issue with another vendors product and no budget... Thanks for all the feedback so far!
Trying to pick the collective brain here! Scenario: We are a growing network service provider with ISP roots. We got into VoIP by selling outsourced hosted PBX services. Merged with a company doing early Asterisk deployment of hosted VoIP. Now doing trunk replacement with Adtran TA9xx at customer premise. PSTN connections are PRI trunks to a wholesale provider routed through ah Atlas 550 to feed our Asterisk and Cisco 5350 SIP gateways, SIP trunking and some offnet via outsources hosted PBX provider. Recently added local ENUM database server feeding the 5350s as dial-peers got unwieldy. Our ultimate goal is to migrate from the outsourced hosted provider to our own Broadsoft platform or equivalent but we're not yet at critical mass to make that financially viable. We're presently peaking at about 150 to 200 simultaneous calls busy hour through our own fabric plus about that on the hosted PBX provider offnet. What I'm looking for: SIP routing engine and/or softswitch but don't need feature server. Needs to do at a minimum: * SIP routing based on destination number or pattern. * Registrar for remote Adtran TA9xx at customer sites. * Interface via SIP with 5350s for calls using PRI to PSTN. * Fax compatibility T.38/G.711 passthrough. * Support multiple SIP carriers inbound and outbound. * CDR generation. * Robust enough for solid performance - failover pair or load-balanced * Ability to grow to handle ~1000 simultaneous calls Nice to have: * LCRE based on time of day or destination pattern. * Geographic redundancy, located in multiple POPs. * NAT traversal B2BUA type functionality * IPv6 or at least a path to IPv6 functionality * Transcoding Suggestions and/or recommendations? -- Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net Impulse Internet Service - http://www.impulse.net/ Your local telephone and internet company - 805 884-6323 - WB6RDV
OpenSIPs/Kamillo/SER or whatever it's called today can do most of this out of the box. I think it's the best option to match your needs if you have the time and engineering to figure it out.
From there you can go to the Acme/Covergence SBC platforms, the lower end Covergence will run in a VM or on your own hardware, but can handle most of what you are looking for.
Only thing on the list this doesn't include is the T.38 fax stuff, but since all of these would require separate media gateways anyway, that's the only place you need the T.38 supported, SER or the SBC would just pass the SDP along. -Scott -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jay Hennigan Sent: Friday, October 30, 2009 4:21 PM To: VoiceOps Subject: [VoiceOps] SIP routing engine - growing pains Trying to pick the collective brain here! Scenario: We are a growing network service provider with ISP roots. We got into VoIP by selling outsourced hosted PBX services. Merged with a company doing early Asterisk deployment of hosted VoIP. Now doing trunk replacement with Adtran TA9xx at customer premise. PSTN connections are PRI trunks to a wholesale provider routed through ah Atlas 550 to feed our Asterisk and Cisco 5350 SIP gateways, SIP trunking and some offnet via outsources hosted PBX provider. Recently added local ENUM database server feeding the 5350s as dial-peers got unwieldy. Our ultimate goal is to migrate from the outsourced hosted provider to our own Broadsoft platform or equivalent but we're not yet at critical mass to make that financially viable. We're presently peaking at about 150 to 200 simultaneous calls busy hour through our own fabric plus about that on the hosted PBX provider offnet. What I'm looking for: SIP routing engine and/or softswitch but don't need feature server. Needs to do at a minimum: * SIP routing based on destination number or pattern. * Registrar for remote Adtran TA9xx at customer sites. * Interface via SIP with 5350s for calls using PRI to PSTN. * Fax compatibility T.38/G.711 passthrough. * Support multiple SIP carriers inbound and outbound. * CDR generation. * Robust enough for solid performance - failover pair or load-balanced * Ability to grow to handle ~1000 simultaneous calls Nice to have: * LCRE based on time of day or destination pattern. * Geographic redundancy, located in multiple POPs. * NAT traversal B2BUA type functionality * IPv6 or at least a path to IPv6 functionality * Transcoding Suggestions and/or recommendations? -- Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net Impulse Internet Service - http://www.impulse.net/ Your local telephone and internet company - 805 884-6323 - WB6RDV _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I would agree with Scott strongly. Scott Berkman wrote:
OpenSIPs/Kamillo/SER or whatever it's called today can do most of this out of the box. I think it's the best option to match your needs if you have the time and engineering to figure it out.
From there you can go to the Acme/Covergence SBC platforms, the lower end Covergence will run in a VM or on your own hardware, but can handle most of what you are looking for.
Only thing on the list this doesn't include is the T.38 fax stuff, but since all of these would require separate media gateways anyway, that's the only place you need the T.38 supported, SER or the SBC would just pass the SDP along.
-Scott
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Jay Hennigan Sent: Friday, October 30, 2009 4:21 PM To: VoiceOps Subject: [VoiceOps] SIP routing engine - growing pains
Trying to pick the collective brain here!
Scenario:
We are a growing network service provider with ISP roots. We got into VoIP by selling outsourced hosted PBX services. Merged with a company doing early Asterisk deployment of hosted VoIP.
Now doing trunk replacement with Adtran TA9xx at customer premise.
PSTN connections are PRI trunks to a wholesale provider routed through ah Atlas 550 to feed our Asterisk and Cisco 5350 SIP gateways, SIP trunking and some offnet via outsources hosted PBX provider. Recently added local ENUM database server feeding the 5350s as dial-peers got unwieldy.
Our ultimate goal is to migrate from the outsourced hosted provider to our own Broadsoft platform or equivalent but we're not yet at critical mass to make that financially viable. We're presently peaking at about 150 to 200 simultaneous calls busy hour through our own fabric plus about that on the hosted PBX provider offnet.
What I'm looking for:
SIP routing engine and/or softswitch but don't need feature server. Needs to do at a minimum:
* SIP routing based on destination number or pattern. * Registrar for remote Adtran TA9xx at customer sites. * Interface via SIP with 5350s for calls using PRI to PSTN. * Fax compatibility T.38/G.711 passthrough. * Support multiple SIP carriers inbound and outbound. * CDR generation. * Robust enough for solid performance - failover pair or load-balanced * Ability to grow to handle ~1000 simultaneous calls
Nice to have:
* LCRE based on time of day or destination pattern. * Geographic redundancy, located in multiple POPs. * NAT traversal B2BUA type functionality * IPv6 or at least a path to IPv6 functionality * Transcoding
Suggestions and/or recommendations?
-- Jay Hennigan - CCIE #7880 - Network Engineering - jay at impulse.net Impulse Internet Service - http://www.impulse.net/ Your local telephone and internet company - 805 884-6323 - WB6RDV _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
-- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671
OpenSIPs/Kamillo/SER or whatever it's called today can do most of this out
of the box. I think it's the best option to match your needs if you have the time and engineering to figure it out.
We started out our voip platform on OpenSIPS + Cisco, and have been pretty pleased with the results. The only thing that is a huge pain is mixing our PSTN upstreams - we have PRIs from the LEC and we have SIP trunks from a LD provider, which, because of their own individual peculiarities, require lots of SIP handholding to make fax, MoIP, etc work reliably. We are very thankful that our prem gear is overwhelmingly Cisco, because consistency makes life better. Our biggest hurdles have been tremendous amounts of development time on OpenSIPS to make it do what we want it to do, and interoperability. Interop - both with our vendors and with random customer PBXs (don't get me start on mismatched NSE/NTE) - has been a major time sink. That said, the bang for the buck is out of this world. Because it's Linux everywhere, load balancing and fail-over is a no-brainer. Need 1000 more calls *right now*? Image whatever server is laying around, make 2-3 LB/DNS changes, and pow, done. Scaling couldn't be easier, imo. That said, we are getting our feet with with OpenSIPS 1.6 B2BUA-style service -- it doesn't proxy RTP streams (can if you want to use MediaProxy addon), but does let it hold up multiple sip dialogs for one call, and also does topo hiding. This turns osips into basically a SBC, without RTP. As long as our RTP interop stays clean, it should be a killer solution. HTH, Randal
Hi All What I need to know: Does the newer PGW 9.8.1 software allow one to configure one C7-Linkset connecting via AS5400XM gateway ?A? to signal for media links that are connecting via a different AS5400XM gateway ?B? Media links on both AS5400XM gateways A & B are connecting to the same ?Media Transfer Point? I know for a fact that previous PGW 2200 versions ie. 9.5.2 could not support this without the use of ?ITP? ( Cisco IP Transfer Point - LinkExtender ). This would simplify configuration and and allow for much less C7 Signaling links I unfortunately cannot test this in the Lab as we do not have enough equipment to simulate this. Thanks Albert
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Hello Albert,
From what you are describing, I see no reason why you cannot achieve what you are after using 9.5(2), we are using 9.5(2) here in both nailed-trunk and call-control modes with both standalone SLT and integrated SLT (AS5400HPX) based signalling links.
(The ITP-L, formally known as the SLT - the product was just renamed, is a different beast to the ITP) Are you saying you have something like the following: GW-A: SS7 Signalling Channel + Trunks using GW-A SS7 link GW-B: Trunks using GW-A SS7 link This should just work - as long as you have defined a C7IPLNK, SESSIONSET, SS7PATH, etc, you can apply the SS7PATH in your trunks file to apply to any IPLNK. I have found no behavioural difference (for signalling purposes) between the Integrated SLT or standalone SLT in our network. I don't have the spare infrastructure in my model at present to re-replicate this scenario, and due to infrastructure relocations the usage of this in our production network has ceased. This documentation, which I assume you have already seen, should cover this scenario: http://www.cisco.com/en/US/docs/ios/12_2t/12_2t11/feature/guide/ftintslt.htm... http://www.cisco.com/en/US/docs/voice_ip_comm/pgw/itp_l/itpl.html#wp1219415 Cheers, Andrew From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Albert Etsebeth Sent: Tuesday, 3 November 2009 12:33 AM To: VoiceOps Subject: [VoiceOps] PGW 2200 version 9.8.1 - C7-Link Hi All What I need to know: Does the newer PGW 9.8.1 software allow one to configure one C7-Linkset connecting via AS5400XM gateway "A" to signal for media links that are connecting via a different AS5400XM gateway "B" Media links on both AS5400XM gateways A & B are connecting to the same "Media Transfer Point" I know for a fact that previous PGW 2200 versions ie. 9.5.2 could not support this without the use of "ITP" ( Cisco IP Transfer Point - LinkExtender ). This would simplify configuration and and allow for much less C7 Signaling links I unfortunately cannot test this in the Lab as we do not have enough equipment to simulate this. Thanks Albert Please note: This email and its content are subject to the disclaimer as displayed at the following link http://www.is.co.za/legal/E-mail+Confidentiality+Notice+and+Disclaimer.htm. Should you not have Web access, send an email to mailto:disclaimers at is.co.za and a copy will be sent to you.
participants (11)
-
abalashov＠evaristesys.com -
acb＠staff.iinet.net.au -
albert.etsebeth＠is.co.za -
carlos＠televolve.com -
jay＠west.net -
jonathan＠thurmantech.com -
lriemer＠bestline.net -
pchilds＠internode.com.au -
rj＠voxeo.com -
rkohutek＠gmail.com -
scott＠sberkman.net