[asterisk-users] Cisco router
I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
Thanks for the info. I didn't have any model in mind, just wondering what was required. Thanks again, much appreciated. Julian 2009/10/14 Jonathan Thurman jthurma...@gmail.com: Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
The lowest end that you can use are 2600, 2600xm, 2800 or 3600. Then like a previous poster said, you need the DSP's and T1/E1 modules, but not all of them support it. NM-HDV2-2T1/E1 are relatively cheap, but you need to make sure that it actually has the t1/ei VWIC in it and it has DSP's in it as well. Some people sell the NM with nothing in it and that is useless. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Wednesday, October 14, 2009 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco router Thanks for the info. I didn't have any model in mind, just wondering what was required. Thanks again, much appreciated. Julian 2009/10/14 Jonathan Thurman jthurma...@gmail.com: Depends on what the router is. If you get a 2800 series router (we use 2801s and 2811s for T1s in production with no major issues). You need the T1/E1 module, DSPs, and an IOS that supports voice. For a 2800 series you would need something like: - VWIC2-MFT-T1/E1 ( or VWIC2-2MFT-T1/E1 if you want two ports) - PVDM2-32 (PVDM2-64 if you want two E1 ports, or two PVDM2-32s) - IOS that supports voice (I use spservicesk9) If you are looking at an older router like a 2651XM or something, you will need something like: - NM-HDV2-2T1/E1 (not the VWIC2, as the NM has build in ports) - PVDM2-32 If you have a specific router in mind, I can be more specific. -Jonathan On Wed, Oct 14, 2009 at 11:00 AM, Julian Lyndon-Smith aster...@dotr.com wrote: I was thinking of putting a cisco router on the E1 line for my test system, so I can have multiple test servers accessing the ISDN, rather than a dedicated server and a TE410 card. I *am* confused at all of the modules for the cisco :) What would be the best router to use to connect 30 channels E1 to SIP ? What modules would I need ? I was going to purchase off ebay as this is purely for testing purposes. TIA ;) Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco router
On Wed, Oct 14, 2009 at 12:57 PM, Julian Lyndon-Smith aster...@dotr.com wrote: Thanks for the info. I didn't have any model in mind, just wondering what was required. If you haven't purchased anything yet, or don't have anything, it might serve you better to look at other products. While the Cisco 2800s that we use work with Asterisk, we use them because that's what we had. I would look at an AudioCodes M1000, or an Adtran 908e or the like. I don't have any experience with E1, but I would guess that there is some support for them by those devices. The AudioCodes is about the same cost as a new Cisco solution, but the Adtran would probably be a lot less. I haven't had a chance to play with Adtran and Asterisk, but you can register at their website and play with all of the CLI / GUIs for all the devices which is really cool. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router for supply a connection from PABX to Asterisk
some howto configuration for asterisk controlling ci$co router (pri/qsig ports especially) using mgcp interests me too... ;-) Yehavi Bourvine +972-8-9489444 wrote I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is a voice bundle of 2,811 + E1 + PDLM card. Note that you need PDLMs as the same number as the PRI channels you are going to define (i.e. 32 PDLMs for each PRI). I am controlling the Cisco via SIP; it works, but a few problems: - Only basic connectivity. No additional features (like names) as the Cisco supports them only via MGCP (in MGCP is passes all the Q.sig signals to the PBX - Asterisk in this case - and it should do all the handling, but I did not find how to do it with Asterisk). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco Router for supply a connection from PABX to Asterisk ?
Hi anyone know if they have a solution in Cisco for: 1- Connect old PABX (with BRI or PRI) to a cisco router 2- Connect this cisco router in SIP to a Asterisk Server I am search if cisco can this and what is the modele for this Thanks ;=) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router for supply a connection from PABX to Asterisk
anyone know if they have a solution in Cisco for: 1- Connect old PABX (with BRI or PRI) to a cisco router 2- Connect this cisco router in SIP to a Asterisk Server I am search if cisco can this and what is the modele for this I am using a Cisco to connect Asterisk via PRI to our Nortel TX-1. The Cisco is a voice bundle of 2,811 + E1 + PDLM card. Note that you need PDLMs as the same number as the PRI channels you are going to define (i.e. 32 PDLMs for each PRI). I am controlling the Cisco via SIP; it works, but a few problems: - Only basic connectivity. No additional features (like names) as the Cisco supports them only via MGCP (in MGCP is passes all the Q.sig signals to the PBX - Asterisk in this case - and it should do all the handling, but I did not find how to do it with Asterisk). - Cisco has no authentication mechanism, so anyone which has access to port 5060 of it can generate calls. Asterisk is not better when it concerned in such sutiations... Regards, __Yehavi: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine.Bruce On 8/23/06, Rich Adamson [EMAIL PROTECTED] wrote: Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I likeusing the TOS header bits personally as lots of other protocols (eg,dns) will eventually match on udp/4569.For the TOS bits v1.2.10, use tos=lowdelay in iax.conf and on thecisco use an access list to match on the tos bits. Something like:access-list 103 permit ip any any dscp cs3access-list 103 permit ip any any dscp efaccess-list 103 permit ip any any tos min-delay= same as tos=lowdelay access-list 103 permit ip any any tos 12For the TOS bits svn truck, the tos= settings have changed inasterisk. Look in the supplied documentation (eg, readme's, sampleconfigs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access listitem dscp cs3 and dscp ef.If you're not all that experienced on cisco qos, then the following isan example of a working config that you should be able to translate into your router config one way or another.class-map match-all voice-rtp match access-group 103class-map match-all www-traffic match access-group 105!policy-map voice-policy class voice-rtp priority percent 40 class www-trafficbandwidth percent 30 class class-defaultfair-queue!interface Dialer0bandwidth 555snip, my specific interface config statements service-policy output voice-policy!access-list 103 permit ip any any dscp cs3access-list 103 permit ip any any dscp efaccess-list 103 permit ip any any tos min-delayaccess-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www anyThe above config provides low-latency priority to voice-rtp, thenprovides an additional qos piece to ensure www-traffic is givenbandwidth before all of the class-default traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% ofbandwidth=555 above) if voice traffic is present. If voice trafficisn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic after the router deals withvoice-rtp traffic. The default class always gets what bandwidth is leftover (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a show access-list 103 from the CLI (onthe router) and watch for matching packets in each access list line.Once you've structured the access list to truly match asterisk traffic, then do a show policy-map interface dialer0 to display how the overallqos structure is functioning.Note that cisco didn't get real serious about IOS qos until v12.2 oftheir IOS code. In v12.2 (and later versions of IOS) there has been asignificant amount of work to bring all of their products into industrystandard implementations / conformance / expectations. If you want toget real serious with cisco's qos stuff, purchase the book End-to-end QoS Network Design and read the 700+ pages devoted to the subject. Itis an excellent book with lots of examples, etc. The book (and actualpractice) suggests IOS v12.3 has more QoS funtionality then v12.2 , andv12.4 has more then v12.3. (The authors of the book back that statementup 100% as well, and they are cisco employees.)In the above config, the bandwidth=555 statement is very important. It should represent the actual outgoing bandwidth for whatever interfaceyou are using and not the theoretical max that someone said you should get.Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time isconsumed sending a non-voice 1500-byte packet, the time is likely to bemore then the 20 millisecond interval for sip/iax traffic. If that is your case, then you may need to forcibly reduce the MTU size of packetsoriginating from other non-voice workstations/servers. The later ciscoIOS versions have a parameter to do that if you can't do it via the workstation/server configuration parameters. If memory serves correctly,that parameter appeared around v12.4 of their IOS.One last item... all of the above deals only with outgoing traffic.You would need to talk to your ISP about QoS for your incoming traffic, and most of the local ISP's don't have a clue. Increasingly, some of thelarger backbone isp's are beginning to understand QoS and some haveactually implemented something. However, those isp's are heading towards providing QoS as a value-add chargeable service (as in MPLS, etc).R.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Cisco Router QOS and IAX2
The majority of the sample qos policies seem to be based on either five or seven qos queues, and most folks don't need all of that. What I've shown as a sample only has three queues; one for voip, one for my outbound web traffic, and the default queue that everything else falls into. You can actually remove the sections relating to web traffic if you don't have a production web server contending for outbound traffic, making it a two-queue policy. R. Bruce Reeves wrote: Thank you so much. After fighting with a large/extensive QOS policy from Cisco's SDM tool, I used your sample and tweaked it for my needs and everything started working fine. Bruce On 8/23/06, *Rich Adamson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I like using the TOS header bits personally as lots of other protocols (eg, dns) will eventually match on udp/4569. For the TOS bits v1.2.10, use tos=lowdelay in iax.conf and on the cisco use an access list to match on the tos bits. Something like: access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay = same as tos=lowdelay access-list 103 permit ip any any tos 12 For the TOS bits svn truck, the tos= settings have changed in asterisk. Look in the supplied documentation (eg, readme's, sample configs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access list item dscp cs3 and dscp ef. If you're not all that experienced on cisco qos, then the following is an example of a working config that you should be able to translate into your router config one way or another. class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue ! interface Dialer0 bandwidth 555 snip, my specific interface config statements service-policy output voice-policy ! access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any The above config provides low-latency priority to voice-rtp, then provides an additional qos piece to ensure www-traffic is given bandwidth before all of the class-default traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of bandwidth=555 above) if voice traffic is present. If voice traffic isn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic after the router deals with voice-rtp traffic. The default class always gets what bandwidth is left over (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a show access-list 103 from the CLI (on the router) and watch for matching packets in each access list line. Once you've structured the access list to truly match asterisk traffic, then do a show policy-map interface dialer0 to display how the overall qos structure is functioning. Note that cisco didn't get real serious about IOS qos until v12.2 of their IOS code. In v12.2 (and later versions of IOS) there has been a significant amount of work to bring all of their products into industry standard implementations / conformance / expectations. If you want to get real serious with cisco's qos stuff, purchase the book End-to-end QoS Network Design and read the 700+ pages devoted to the subject. It is an excellent book with lots of examples, etc. The book (and actual practice) suggests IOS v12.3 has more QoS funtionality then v12.2 , and v12.4 has more then v12.3. (The authors of the book back that statement up 100% as well, and they are cisco employees.) In the above config, the bandwidth=555 statement is very important. It should represent the actual outgoing bandwidth for whatever interface you are using and not the theoretical max that someone said you should get. Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time is consumed sending a non-voice 1500-byte packet, the time is likely to be more then the 20 millisecond interval for
[asterisk-users] Cisco Router QOS and IAX2
I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. -- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
On 8/23/06, Bruce Reeves [EMAIL PROTECTED] wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. Can't you just setup a policy class based on the host/UDP ports participating in your IAX networking? The RTP isn't separated in IAX so you don't need to keep track of signalling and RTP traffic separately like you would with SIP. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce, this might be able to help give you some hints or a place to start: http://www.voip-info.org/wiki/view/QoS+Cisco Hope that helps \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. I just spent some time doing this myself. If your routers already can prioritize traffic based on the TOS bits in your IP traffic, setting the tos in iax.conf should be the first place to start. If your routers don't do that already you will need to mess with setting up queuing/policies on the routers. Here are some places to start: Here are some awesome descriptions on how QoS works and what the different methods of implementing queuing and traffic shaping on Cisco hardware are. They may be a bit dated depending on what kind or routers you are using: http://www.netcraftsmen.net/welcher/papers/qos1.html http://www.netcraftsmen.net/welcher/papers/qos2.html http://www.netcraftsmen.net/welcher/papers/qos3.html Here are some pages from the wiki that talk about QoS on cisco hardware. Not sure what type of queueing this uses, but it allocates a certain amount of available traffic to voice traffic. Any unused voice traffic will be shared by what's left: http://www.voip-info.org/wiki/view/QoS+Cisco This page seems to talk about CAR (Committed Access Rate): http://www.voip-info.org/wiki/view/QoS+Cisco+IOS I ended up using priority queuing on my routers, giving voice first priority and everything else lower priorities. Not the best solution but it was the easiest for me to implement with the versions of IOS I have. If you would like to see my configs, let me know. -Dave Fullerton ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco Router QOS and IAX2
Bruce Reeves wrote: I'm needing some pointers from anyone who has been able to get Cisco routers to recognize the iax protocol and perform QOS on it. Or if there is a better way to get my iax traffic prioritized by the router. You can either match on udp/4569, or, match on TOS header bits. I like using the TOS header bits personally as lots of other protocols (eg, dns) will eventually match on udp/4569. For the TOS bits v1.2.10, use tos=lowdelay in iax.conf and on the cisco use an access list to match on the tos bits. Something like: access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay = same as tos=lowdelay access-list 103 permit ip any any tos 12 For the TOS bits svn truck, the tos= settings have changed in asterisk. Look in the supplied documentation (eg, readme's, sample configs) for exactly what is allowed in terms of DiffServ (new term for TOS basically). You'll find examples that support the above access list item dscp cs3 and dscp ef. If you're not all that experienced on cisco qos, then the following is an example of a working config that you should be able to translate into your router config one way or another. class-map match-all voice-rtp match access-group 103 class-map match-all www-traffic match access-group 105 ! policy-map voice-policy class voice-rtp priority percent 40 class www-traffic bandwidth percent 30 class class-default fair-queue ! interface Dialer0 bandwidth 555 snip, my specific interface config statements service-policy output voice-policy ! access-list 103 permit ip any any dscp cs3 access-list 103 permit ip any any dscp ef access-list 103 permit ip any any tos min-delay access-list 103 permit ip any any tos 12 access-list 105 permit tcp any eq www any The above config provides low-latency priority to voice-rtp, then provides an additional qos piece to ensure www-traffic is given bandwidth before all of the class-default traffic. In other words, voice-rtp traffic will always get 40% of the bandwidth (eg, 40% of bandwidth=555 above) if voice traffic is present. If voice traffic isn't present, that bandwidth can be used by other qos sections or by the default class. Same with www-traffic after the router deals with voice-rtp traffic. The default class always gets what bandwidth is left over (or all bandwidth if there is no voice-rtp or www-traffic). To troubleshoot the above, do a show access-list 103 from the CLI (on the router) and watch for matching packets in each access list line. Once you've structured the access list to truly match asterisk traffic, then do a show policy-map interface dialer0 to display how the overall qos structure is functioning. Note that cisco didn't get real serious about IOS qos until v12.2 of their IOS code. In v12.2 (and later versions of IOS) there has been a significant amount of work to bring all of their products into industry standard implementations / conformance / expectations. If you want to get real serious with cisco's qos stuff, purchase the book End-to-end QoS Network Design and read the 700+ pages devoted to the subject. It is an excellent book with lots of examples, etc. The book (and actual practice) suggests IOS v12.3 has more QoS funtionality then v12.2, and v12.4 has more then v12.3. (The authors of the book back that statement up 100% as well, and they are cisco employees.) In the above config, the bandwidth=555 statement is very important. It should represent the actual outgoing bandwidth for whatever interface you are using and not the theoretical max that someone said you should get. Also note that for relatively slow speed interfaces (eg, most dsl's), the outgoing bandwidth is rather slow. If you calculate how much time is consumed sending a non-voice 1500-byte packet, the time is likely to be more then the 20 millisecond interval for sip/iax traffic. If that is your case, then you may need to forcibly reduce the MTU size of packets originating from other non-voice workstations/servers. The later cisco IOS versions have a parameter to do that if you can't do it via the workstation/server configuration parameters. If memory serves correctly, that parameter appeared around v12.4 of their IOS. One last item... all of the above deals only with outgoing traffic. You would need to talk to your ISP about QoS for your incoming traffic, and most of the local ISP's don't have a clue. Increasingly, some of the larger backbone isp's are beginning to understand QoS and some have actually implemented something. However, those isp's are heading towards providing QoS as a value-add chargeable service (as in MPLS, etc). R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco router and 488 Not acceptable here messages
Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting 488 Not acceptable here messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting chan_sip.c:3434 process_sdp: Insufficient information for SDP (m = '', c = '') messages, which is strange because the m and c attributes are definitely there. When I looked closer, they are received on the first INVITE, then asterisk says 'proxy auth required', but the INVITE packet with the proxy auth doesn't have the attributes. A tcpdump on the asterisk server confirms this. But, when I do a ethereal dump on my PC where I'm running SJphone, the attributes are there in the packet. So something is futzing with my packets, and screwing them up. There is a Cisco router on my end of the link, so I'm suspecting that! Any suggestions? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco router and 488 Not acceptable here messages
Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Monday, 12 June 2006 00:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Cisco router and 488 Not acceptable here messages Are there any known problems with Cisco routers (Cisco 837) and SIP sessions? I have been trying to track down a problem for about 3 hours now and I think the Cisco router is the culprit!!! I keep getting 488 Not acceptable here messages, which are apparently normally the message you get when a common codec can't be found. I'm also getting chan_sip.c:3434 process_sdp: Insufficient information for SDP (m = '', c = '') messages, which is strange because the m and c attributes are definitely there. When I looked closer, they are received on the first INVITE, then asterisk says 'proxy auth required', but the INVITE packet with the proxy auth doesn't have the attributes. A tcpdump on the asterisk server confirms this. But, when I do a ethereal dump on my PC where I'm running SJphone, the attributes are there in the packet. So something is futzing with my packets, and screwing them up. There is a Cisco router on my end of the link, so I'm suspecting that! Any suggestions? Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco router and 488 Not acceptable here messages
On Jun 11, 2006, at 8:15 AM, James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. If you are behind a NAT perhaps two SIP devices are both trying to use 5060? Just a thought. Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco router and 488 Not acceptable here messages
James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. James You probably have are behind NAT and your NAT device has a SIP ALG. Changing the port disables the ALG. The ALG is broken. -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco router and 488 Not acceptable heremessages
On Jun 11, 2006, at 8:15 AM, James Harper wrote: Additionally, just to satisfy myself that I wasn't going mad I changed the port from 5060 to 5070 and now things are working, so something is definitely playing up on port 5060. If you are behind a NAT perhaps two SIP devices are both trying to use 5060? Packets are getting out, but one critical packet has had the body of the INVITE removed, so Asterisk at the other end thinks that no audio codecs have been proposed. James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Router FXO / Skinny
Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. www.kingston.com has inexpensive DRAM and FLASH upgrades for Cisco (and many other devices). For my 1750 I upgraded the DRAM, but not the flash. I tell the router to load it's IOS via TFTP from a local server. It works reasonably well. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Router FXO / Skinny
Hey, Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. Does anyone know how to configure this 2600 with Asterisk? Thanks, Erik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco Router FXO / Skinny
Does anyone know how to use the cisco router with an fxo wic with Asterisk? I don't have enough space on this device to support an IOS that supports sip or h323. Currently the only one signaling in there says Cisco. I assume this is the skinny protocol. This is Cisco's pre-SIP implementation of VoIP. Outside of a Cisco environment it is not very useful. I would suggest trying to get a later image on the box. You can always boot it using tftp if your flash filesystem is not big enough (ie 'boot system tftp xx'). If system memory is your problem try a later IP+ image or pull one of your cards. Does anyone know how to configure this 2600 with Asterisk? Maybe someone else can offer their config. There are a few Cisco-Asterisk users here. It would be nice to see a WIKI page on ISDN/PSTN-CISCO-Asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco router *
Hi all, Just checking that what I want to do is possible. I've got a cisco 3620 in my lab (IP plus ios). I'm thinking of moving my current pots lines to isdn BRI at home. What I'm thinking of is putting a wic-1b(s/t) into the 3620 and using that to pass incoming calls to * via sip. As I understand it this should be doable. I suppose my question is will the 3620 pay all caller ID information to * so that it can be logged. I don't yet have the BRI installed, so I can't post configs etc. TIA. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cisco router *
Hello, On an AS5350 this works so I expect this to work too your 3620 Regards, Niels -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence Sent: Tuesday, October 26, 2004 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] cisco router * Hi all, Just checking that what I want to do is possible. I've got a cisco 3620 in my lab (IP plus ios). I'm thinking of moving my current pots lines to isdn BRI at home. What I'm thinking of is putting a wic-1b(s/t) into the 3620 and using that to pass incoming calls to * via sip. As I understand it this should be doable. I suppose my question is will the 3620 pay all caller ID information to * so that it can be logged. I don't yet have the BRI installed, so I can't post configs etc. TIA. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco router *
On Tue, 2004-10-26 at 08:02, Jon Lawrence wrote: Hi all, Just checking that what I want to do is possible. I've got a cisco 3620 in my lab (IP plus ios). I'm thinking of moving my current pots lines to isdn BRI at home. What I'm thinking of is putting a wic-1b(s/t) into the 3620 and using that to pass incoming calls to * via sip. As I understand it this should be doable. I suppose my question is will the 3620 pay all caller ID information to * so that it can be logged. Caller-ID is a POTS only feature, it isn't part of the ISDN call setup. What *is* part of the ISDN call setup is the originating and destination numbers. This info provides caller-id like features and is what gets converted to real caller-id on POTS lines. This is also why caller-id blocks don't work on ISDN lines. You can't block the call setup data on ISDN. If you always want to know who's calling no matter what, use ISDN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco router *
Won't work. The WIC-1B is a data only. You need an NM-1V (or NM-2V) and a VIC-2BRI to terminate voice and pass it via IP. Peder Jon Lawrence wrote: Hi all, Just checking that what I want to do is possible. I've got a cisco 3620 in my lab (IP plus ios). I'm thinking of moving my current pots lines to isdn BRI at home. What I'm thinking of is putting a wic-1b(s/t) into the 3620 and using that to pass incoming calls to * via sip. As I understand it this should be doable. I suppose my question is will the 3620 pay all caller ID information to * so that it can be logged. I don't yet have the BRI installed, so I can't post configs etc. TIA. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco router *
On Tuesday 26 October 2004 14:13, [EMAIL PROTECTED] wrote: Won't work. The WIC-1B is a data only. You need an NM-1V (or NM-2V) and a VIC-2BRI to terminate voice and pass it via IP. Should have known Cisco would want more money from me to get this working :) It'll probably be cheaper to get a 1751V and a vic-2bri rather than a nm-2v + vic-2bri for the 3620. Going to have to do some more saving up. Plan 2. What's the best bri card to put directly in a * box in the UK. If anyone would like to quote for a NM-2V + vic-2BRI please send off list. I'm assuming that this is about the only way to get a 3620 connected to a BRI for voice ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CISCO router closes the connection before starting conversation
I have a router Cisco 2650XM with a VIC 2ISDN-BRI card that connects to a traditional PBX. I have achived to pass calls from * to the PBX but I failed the reverse. When a place a call to telephone via the PBX, it passes to Cisco router and then * is used to make the call to a Xlite SIP Softphone. The Xlite is ringing once and the call is terminated. By examining the logs and with SIP Debug I found that exactly at the moment the Xlite starts ringing a CANCEL command comes from the cisco router. Any comments? Thanks in advance for anyone who takes the time to reply Savas Pavlidis begin:vcard fn:Savas Pavlidis n:Pavlidis;Savas email;internet:[EMAIL PROTECTED] tel;work:+30 2310 573300 tel;fax:+30 2310 752280 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco Router as Gateway for FXO Ports
Because of all the troubles I am having with the TDM400P cards working consistently for outgoing calls, I am trying to setup a Cisco router as my pstn gateway. The router is a 1751-V with a VIC2-4FXO card installed. I followed the examples at the following two links to get it setup: http://www.voip-info.org/wiki-Asterisk+cisco+FXO http://www.tape.net/~gerry/asterisk/cisco26x0.html After everything is up and running, outgoing calls from my Asterisk PBX route just fine through the Cisco without any trouble but incoming calls to the Cisco are picked up and then immediately given a busy signal. Is anyone successfully getting this type of setup to work? Any help would be greatly appreciated! Thank you. Bryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco router/SIP gateway registration
Is there a way to have a Cisco SIP gateway register with Asterisk? The current setup just drops calls into the sip.conf default context which works fine but has some security risks since anyone who can install XTEN and has access to my LAN can then use this context to drop calls in I'd like to be able to get inbound calls from the cisco in a from_gw context, then I can just set the default context to a simple Congestion dialplan... Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
