Re: [Asterisk-Users] TDMoE and Badness in Kernel
On Fri, 2005-10-21 at 09:26 -0700, trixter aka Bret McDanel wrote: On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote: I received some postings back, as I was trying to do the same thing. it' is a problem with Kernel 2.6... 2.4 works fine .. this is the summary I got from reading the posts before. I hope that helps... I dont have the ability to go DOWn in kernel to 2.4.. the wiki suggested that it was a problem with softirq.c in the kernel and that this was fixed at some point. What 2.6 version are you running that you have this problem? I've seen this on just about everything 2.6.9 and above and up to 2.6.13. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voice over atlantic
Hi David, I just looked at my iax.conf on one of my boxes in Argentina and actually there are no jitterbuffer settings indicated so I'm assuming it is using Asterisk defaults. We are experimenting with G.729 on these IAX trunks also and I just realized I have no accurate means of measuring bandwidth consumption vis-a-vis GSM/G.729. I think I'll pose that question to the group in another message to see what recommendations and best practices are out there. Or, do some research. Best of luck. On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voice over atlantic
Forgot the version: Asterisk 1.0.7 On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
On Thu, 2005-09-08 at 16:26 +0200, Simone Cittadini wrote: Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? You might want to offload some of that PRI termination to an external device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone fonebridge device and then trunk it to your Asterisk servers. But putting more then 2 quad cards in a single server is not safe. 1 per server would be more acceptable. This link might be helpful to you: http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large Good luck. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voice over atlantic
David, I have IAX trunks running between the US and S. America using the GSM codec and jitterbuffer=yes and the quality seems very good to my ears. Don't have the details of the jitterbuffer parameters right now but hopefully this will give you some useful feedback. Good luck. On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote: Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: equipment configuration help
That is correct. Normally the layer 3 switches include advanced features such as QoS but they may be available on simpler layer 2 switches. I think the key words to look for are 'Managed, QoS (802.1p) with priority queues, VLAN, (802.1q)'...maybe even PoE if you go with some SIP phones in the future that can be powered by Power Over Ethernet. Something else to keep in mind. best of luck. On Thu, 2005-09-01 at 22:03 -0500, Erick Perez wrote: Why an L3? just for the QoS part? I checked the alliedtelesyn 8624T at $1000.00 http://www.cdw.com/shop/products/default.aspx?EDC=772793 but i also looked at the 8550T which has 48 port 10-100 but L2 http://www.cdw.com/shop/products/default.aspx?EDC=773964RecommendedForEDC=772793RecoType=upsell at 900.00 is the QoS different? sorry for the question but i keep reading that asterisk needs qos to function better. Thanks, On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick- Can't say if they will or not. In theory they should respect all outgoing traffic unless being filtered by another device such as your PIX. You might want to check with the ADSL router manufacturer just to be safe. On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote: Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2
Re: [Asterisk-Users] Re: equipment configuration help
Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have to buy an analog card? or maybe link me to a web site that explains (with images) how a t1/e1 is managed? Thanks, and I apologize for this completely newbie question. I've never seen images or instructions on how to handle this. Im not even sure im using the right terms in Google. -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: equipment configuration help
Erick- Can't say if they will or not. In theory they should respect all outgoing traffic unless being filtered by another device such as your PIX. You might want to check with the ADSL router manufacturer just to be safe. On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote: Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have
Re: [Asterisk-Users] Re: equipment configuration help
Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel. That way you'll be able to maintain your proprietary Nortel phones and won't need a channel bank. Your implementation would be something like this: Cable Worthless E1(or whoever)--Asterisk(Sangoma port 1)-(Sangoma port 2)--Meridian---Nortel Digital phones suerte, -M On Wed, 2005-08-31 at 18:37 -0500, Erick Perez wrote: Update to myself: So in terms of equipment I will need: Sangoma a102 E1 (two E1 ports) plus a E1 crossover cable a channel bank with 8 FXS ports sounds expensive for just 8 analog ports. Any ideas? On 8/31/05, Erick Perez [EMAIL PROTECTED] wrote: Hi, Im about to start shopping parts for an * box. We are migrating from a Meridian Norstar+ Modular ICS Here are the customer details: a) Meridian with 8 analog lines card and 32 nortel digital phones and voicemail. We will interface * to the meridian using the analog ports so we dont loose the phones. b)half E1. The * box will get half E1 (with DID) for connecting to the local telco. We need two recepcionist/operator phones (sip or whatever) So in terms of equipment I will need: Sangoma a101 E1/PCI an 8 port analog card a channel bank? Can someone tell me if i really have to buy an analog card? or maybe link me to a web site that explains (with images) how a t1/e1 is managed? Thanks, and I apologize for this completely newbie question. I've never seen images or instructions on how to handle this. Im not even sure im using the right terms in Google. -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
