Re: [asterisk-users] need to find firmware for cisco ata-188
Actually no. But i cannot get a smartnet on an ATA-188. At least not in latinamerica. Actually, all ata-188/186 come with sccp, i just reflashed mine to sip and now i want it back to sccp. it was very dissapointing to learn that i cannot download *any* sccp firmware, not even the original one. Any other suggestions? On Tue, Oct 27, 2009 at 6:53 PM, Steve Howes st...@geekinter.net wrote: On 27 Oct 2009, at 23:29, Erick Perez wrote: any links beside cisco to download the firmware? i do not have a valid contract, so cisco does not allow me to download it. So you want to pirate it instead? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] need to find firmware for cisco ata-188
Hi there, I have an old Cisco ATA-188-I2-A that I want to revive but with SCCP (right now it has SIP). the version i am looking for is ata_03_02_04_sccp_090202_a.zip i want to do a home experiment with chan_sccp and some recompilations any links beside cisco to download the firmware? i do not have a valid contract, so cisco does not allow me to download it. thanks in advance. erick. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
I am fairly certain he was simply reporting the results (for posterity) of the event having already happened. Good to know (I guess?) that such small hardware can acheive the performance that was squeezed out of it. Impressive. All THAT said, I am unconvinced that there was no sales effort involved in sending out millions of unsolicited calls. Claim if you like that this was some public information event (which you fail to expand much upon) and convict me of mistrust, but who would have paid for such a thing. TV ads, radio spots, billboards, etc., are much more effective for public information. Unsolicited calls on that order mean only one thing to me - SPAM. So what wonderful product were you informing the public about with regard to the looming threat of illness? Jeff, indeed i was posting for posterity. Maybe someone will benefit in an outbound-only scenario that he/she will not need a supercomputer to pump a 20sec audio clip. Again, this was a public service. And indeed TV and radio was used. Unless you live in a bubble, you may have heard about AH1N1 virus. Which unfortunately hit us (Panama, Republic of Panama, Central America) very hard. I foud very repetitive to tell in my posts that i am from panama, central america, blah,blah blah. Anyways, a quick google search of this forum will also revealed that i am kind of a regular poster and even my cellphone is listed here (Jon Pounder, my cellphone is +507 6675 5083 in case YOU want to sell me a car loan, i dont mind getting a call. Im a IT consultant and i have a chargeback line. Please call me as many times as you want...please do so between 10pm and 6am where my chargeback is the most expensive). Guys, Grow up! Next time someone needs to learn mouth-to-mouth and CPR lessons, please DONT teach him. Because, following your inmature way of thinking, the person who wants to learn CPR may as well be looking for information to learn how to suffocate people. Next time your son wants to know how gasoline works or how is being produced. Please keep your familiy in ignorance. You may be training the next crazy person who will burn things all around the world. But, you wont do that, do you? Again, I always tell my familiy that keeping others in ignorance is bad. but sometimes it must be done for the sake of a greater good, and my comment is always followed with good and sound examples (atomic technology, viruses, etc). But I forgot that Asterisk, the phone lines and a calling system is the way the world is going to be dominated by the martians. So the secret about phone system calculations must be keept in Area 51. Now I understand Kevin Mitnick. Cheers to all. Bye. Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 400 calls at g711 how much cpu power
I totally agree with you Jeff, however some of us do not actually sell viagra over the phone. This is a campaign to spread a message to the population about the health prevention steps that should be taken in order to prevent diseases that are affecting our population. I do understand all of you to be reluctant to help with this post. However judging before listening has been the most devastating problem humans have. We simply do not trust each other. However, just for the sake of posterity: Hardware/Software just one server Dell 2950 / 4GB RAM / four 72Gb ultra320 SCSI hard disks built as RAID-0 Debian as the OS (in 32 bit mode) Asterisk 32 bit 1.4 compiled manually (codecs removed, modules removed,etc, a ton of pure CRAP out!) Only g711/SIP was used 20 second clip was served from ramdisk Dialer: SmoothTorque (those guys simply ROCK!)( setup outbound mode ONLY!) Network: 50 Mbit fiber link to telco provider. Pure IP, no QoS. We were pumping 3k calls-setup/second to the session controller at telco's side. Until we reached controller's max of 10k calls. Server load was NEVER above 3.2 thanks to all for your help. On Thu, Apr 2, 2009 at 7:36 PM, Jon Pounder j...@inline.net wrote: Erick, how about posting your home phone number here so we can all call you and play a 20second audio clip - I am sure you would see nothing wrong with that would you ? ContactTel Business wrote: Your right, i don't think we would help someone asking on advice to send 1 million emails for Viagra would we ? So why the hell aren't we thinking straight and tell the poor guy? Ive seen dialer app that where legit, even worked on some for the military. But this is just spam /pham (phone spam) send 10USD to my email ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-02-09 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 400 calls at g711 how much cpu power
We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions
Hi all, thanks for the excellent information about the banks and usb banks. some tech details will prevent us from using usb units. The trunks will be 500 feet away from the new location of the ip-pbx so we have decided to go with channel banks for the trunks and sending the E1 signal over cat 5 (E1 signal can travel un-repeated over 5000 feet) So far we are reading/evaluating about rhino channel banks and a quad E1/T1 (pci-e) on the asterisk box. thanks again -- Erick Perez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk across a firewall
On Wed, Feb 11, 2009 at 1:56 PM, Gordon Henderson gordon+aster...@drogon.net wrote: On Wed, 11 Feb 2009, Erick Perez wrote: Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big mistake. my customer has a dumb firewall (not SIP aware) that will not replace. he wants another box to do the magic. I have many customers like that, and working from home is gaining momenting where I live... So the scenario (if I interpret it correctly): Asterisk at HQ is behind a NAT firewall with remote users (who themselves may be behing a NAT firewall) HQ needs a static IP address on the outside and plenty of bandwidth. The dumb router at HQ needs to port-forward external port 5060 and 1-2 into the asterisk box (you can limit this range - see rtp.conf) Most dumb routers can port-forward. Asterisk needs to know it's LAN and extneral ip address - sip.conf, externip= and localnet= remote extensions need nat=yes in sip.conf and that's basically it. If the remote extensions are themselves behind a NAT firewall, then the easiest way to get them through it is by using a stun server - ether run your own, or use someone elses... Do not do any port-forwarding at the remote users sites. Yes, you can fiddle about with proxies, gateways, etc. but keep it simple to start with and I have many installations doing it this way and it just works. One day I'm sure I'll trip up, but until then... Pitfalls - the same with all VoIP - bandwidth, espeically outgoing b/w from HQ. Broken NAT gateways, and routers which have SIP ALGs built in which are also broken. (Turn them off!) Routers with broken SIP ALG are the biggest PITA to work round. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thank you all for the excellent responses. I will do some test here to decide on a method/technology to use. -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk across a firewall
Excuse my ignorance but if i have an asterisk in a LAN, and i have users in their homes/internet (dozens), in order to correctly connect those users across my firewall, what is the technology that i need to buy, called? secure border gateway? session controller? secure gateway? the audiocodes site seems to have many names for the same thing...but i better ask here and learn before i make a big mistake. my customer has a dumb firewall (not SIP aware) that will not replace. he wants another box to do the magic. -- Erick Perez Cel +(507) 6675-5083 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connecting 66 analog phones to asterisk - hardware suggestions
Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with other units on the astribank, as well as to the pbx, pstn, etc? The asterisk pbx will be a server-class Hp Proliant unit (potentially a dl320). i must make sure the astribanks will not die when fully utilized. other hardware suggestions for this task will be nice. thanks, -- Erick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] arris tm502g cablemodem FXS ports and zaptel 1.4.8
Hi there, I have a cablemodem, ARRIS brand, model tm502G. It has two FXS ports. I was wondering if anyone has details about the correct signalling of these FXS ports when connected to original X100p. Tests: fxsks on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing but the zap channel sees no call coming in. fxsls on the zapata.conf and zaptel.conf files. From my cellphone I call the ARRIS, it starts ringing, zap channel picks up the call. all good. fxsgs on the zapata.conf and zaptel.conf files. ztcfg reports error about invalid mode. Well, I used loopstart as the signal, however when using it I face one very nasty issue. My asterisk/zap channel does not detect hangups correctly. I have enabled busydetect but it's kind of unreliable. Specially when using DISA, if one of my external callers use DISA and the external caller hangsup, asteirsk wont see athing and will keep both zap channels open. I will like some suggestions with this as i am not sure if it's related to signalling in the ARRIS or maybe some tweaking i can do in the x100p (true x100p). Thanks, -- Erick Perez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] autoprovision 200+ linksys phones setup
Hi there, We have plans to install an office (not call center) with the following setup: 200 linksys 942 phones (sip + g711) on a LAN a server with a dual port E1 sangoma and a remora card with 4 fxo modules. So far when we want to setup a linksys phone, we need to use the http interface of each phone, disable/enable a lot of things and plug it into the network. this is not the best scenario for us but im sure there must be something we can do to speed things up. We are looking into a distribution (freepbx or pure asterisk,or something else) with links to documentation to enable autoprovisioning on the linksys phones. What we want to achieve is enabling the linksys phones to be plugged into the lan, grab a configuration from tftp or http and be assigned the next free extension. (fonality does something like that with polycoms) So far, the autoprovisioning links i've found talk about polycom phones and grandstream. but in this office (and country) linksys is better to get and much less expensive than polycom phones. Maybe some distro i haven't checked out that autoprovisions linksys 942? also, a guidance (howto, manual, web link) on autoprovisioning will be gladly welcomed. Thanks, -- Erick Perez ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7940G licensing with asterisk
Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, -- Erick Perez ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can handle the 7940G ? The 7941G does conform to the standard but it only support SCCP (shame on cisco). On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
As it turns out the telco was not routing the calls to us, a little misktake they said after 3 days of being with no service. The line was not CAS, it was CCS, no need to compile unicall. Whatever they meant with your card has to be configured with DSS1 will remain in mystery. Maybe someone here can tell me what they mean. The configuration I previously listed is valid for lines in Panama City, Panama. With the telco being Cable Wireless Panama and the asterisk with a sangoma A102. If there's any Cable wireless tech reading this. Guys, your support s*cks big time. Thanks to all for your kind and prompt help. On 7/28/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: If you do not have any alarms and PRI debug span 1 still gives you nothing then you need to call your telco and say I'm not getting any Q.931 messages on the D-Channel. Stephen Bosch wrote: Erick Perez wrote: Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. Hmn, well, that's telling. Are you using the correct cable? Is the cable plugged into the correct port on the card? The 102 is a two-port. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected
Yes I do. I even did a pri debug span 1 and when I call the asterisk box, it sees nothing. On 7/26/07, Idris AVCI [EMAIL PROTECTED] wrote: Do you have any extension in default context of your extensions.conf file to accept incoming calls ? It must be something like; exten = 12345678,1,Answer() exten = 12345678,2,Playback(Welcome) ... 12345678 = The DID number you are calling to reach E1 Idris -Original Message- From: Erick Perez [mailto:[EMAIL PROTECTED] Sent: Thursday, July 26, 2007 7:03 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming calldetected Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help with mfcr2 and pri
Hi, While I wait for my unresponsive telco to provide some assistance, can you provide some configuration details for the following config? Sangoma 102 (dual E1) card Location: Panama, Central America Telco: Cable Wireless Panama Lastest stable asterisk 1.2.x compiled from sources Site A in one office Site B is another office in another town When I asked the telco about using CAS or CCS and CRC4 or NCRC4 the technician said: what? im not sure what you mean. Normally it should be CAS/NCRC4 with an E1 MFCR2 right? and CCS/NCRC4 with Euro ISDN PRI on E1 right? What stream are you going to use (structured/unstructured) structured G 703; TS 16: Signalling Line core (HDB3/AMI) HDB-3 Leased line length (wireline of G703 trunk) G.SDHSL Channel level protocol(Site a) MFC-R2 Channel level protocol(Site b) Euro ISDN PRI How should I configure my sangoma with this settings? zaptel and zapata? what of the many unicall downloadables should I use? any other questions I should ask to my telco? Thanks, -- Erick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help with mfcr2 and pri
I have received the follwing info from my telco. E1, PRI, CAS, HDB3, dss1 any help? On 7/25/07, Erick Perez [EMAIL PROTECTED] wrote: Hi, While I wait for my unresponsive telco to provide some assistance, can you provide some configuration details for the following config? Sangoma 102 (dual E1) card Location: Panama, Central America Telco: Cable Wireless Panama Lastest stable asterisk 1.2.x compiled from sources Site A in one office Site B is another office in another town When I asked the telco about using CAS or CCS and CRC4 or NCRC4 the technician said: what? im not sure what you mean. Normally it should be CAS/NCRC4 with an E1 MFCR2 right? and CCS/NCRC4 with Euro ISDN PRI on E1 right? What stream are you going to use (structured/unstructured) structured G 703; TS 16: Signalling Line core (HDB3/AMI) HDB-3 Leased line length (wireline of G703 trunk) G.SDHSL Channel level protocol(Site a) MFC-R2 Channel level protocol(Site b) Euro ISDN PRI How should I configure my sangoma with this settings? zaptel and zapata? what of the many unicall downloadables should I use? any other questions I should ask to my telco? Thanks, -- Erick. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.23 and Sangoma a102 no incoming call detected
Hi, after many issues we finally managed to make our system do outgoing calls with perfect quality. However I cannot detect *any* form of incoming call. when I use an outside phone to call the E1 connected to the sangoma a102, I instantly get a fast busy tone. My /etc/zaptel.conf is: loadzone=us defaultzone=us #Sangoma A102 port 1 [slot:1 bus:4 span: 1] span=1,0,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 My /etc/asterisk/zapata.conf is: [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no #include zapata-auto.conf Zapata-auto.conf has: callerid=asreceived ;Sangoma A102 port 1 [slot:1 bus:4 span: 1] switchtype=euroisdn context=from-pstn group=0 signalling=pri_cpe channel = 1-15,17-31 Note: According to the tech support in the local telco, my E1 should be: E1 PRI, CAS, HDB3, NCRC4, DSS1 However if I configure the card for CAS, it will never connect. My card is currently configured (and makes only outgoing calls) as: E1 PRI, CCS, HDB3,NCRC4 (i have no idea what dss1 is or where it goes) My /etc/wanpipe/wanpipe1.conf is: [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 1 PCIBUS = 4 FE_MEDIA= E1 FE_LCODE= HDB3 FE_FRAME= NCRC4 FE_LINE = 1 TE_CLOCK= NORMAL TE_REF_CLOCK= 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE= NO LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES thanks for your help. -- Erick Perez ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queuemetrics and Asterisknow
Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez ___ --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] Re: Queuemetrics and Asterisknow
I realized that queuemetrics uses Java. Is java available as an rpath package or do I need to get it from sun? Also, will it break asterisknow? Thanks. On 5/21/07, Erick Perez [EMAIL PROTECTED] wrote: Can I use queuemetrics with asterisknow? I mean, if I modify the dialplan to use queuemetrics (I still don't know if it's possible), will I loose my changes when the time comes to do a conary update of the asterisknow package? thanks, -- Erick Perez -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisknow b5 - trouble registering at voip provider
Hi, there. I have asterisknow beta 5 with the following data: Ip 192.168.0.60 mask 255.255.255.0 gw 192.168.0.1 the router (a linksys) has port forwarded the port udp 5060 and from 16384 to 16482 udp-tcp from the internet to the asterisk machine. the only protocol allowed is g729. Which work fine for the ip phones I already have setup in the LAN. My problem is trying to register to a voip provider. in the asterisknow gui I provide: protocol sip register (checked) host sf2.clarocom.net username (my phone number) password (assigned password) While executing sip show claro91 asterisk*CLI sip show peer claro91 asterisk*CLI * Name : claro91 Secret : Set MD5Secret: Not set Context : DID_ Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: 1 Pickupgroup : 1 Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : 2029191 MaxCallBR: 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID: No Subscriptions: No Overlap dial : No DTMFmode : auto LastMsg : 0 ToHost : sf2.clarocom.net Addr-IP : 200.105.69.132 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 2029191 SIP Options : (none) Codecs : 0x80100 (g729|h263) Codec Order : (g729:20) Auto-Framing: No Status : Unmonitored Useragent: Reg. Contact : asterisk*CLI asterisk*CLI and when i try to call with my lan phones to the outside via the claro91 trunk, I get asterisk*CLI -- Executing [EMAIL PROTECTED]:1] Macro(SIP/6000-0820e870, trunkdial|SIP/claro91/66944780) in new stack -- Executing [EMAIL PROTECTED]:1] Dial(SIP/6000-0820e870, SIP/claro91/66944780) in new stack -- Called claro91/66944780 [May 13 17:37:40] WARNING[5522]: chan_sip.c:11860 handle_response_invite: Received response: Forbidden from 'Erick Perez sip:[EMAIL PROTECTED];tag=as7eabcb2e' -- SIP/claro91-082127d8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/6000-0820e870, s-CONGESTION|1) in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/6000-0820e870, ) in new stack == Auto fallthrough, channel 'SIP/6000-0820e870' status is 'CONGESTION' asterisk*CLI If I switch from my asterisknow box to the linksys box (that has two rj11 ports) then the registration is fine. I would like some guidance as to how to properly format the registration string for my provider. thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk 1.4 and Cisco Phones 7940
I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- Erick Perez ___ --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] Issues with a Linksys SPA 2102 and asterisk
where to change packet size? On 3/9/07, Luki [EMAIL PROTECTED] wrote: Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? They work fine with Asterisk; most likely it's your wireless link that's the cause of your problem. The jitter buffer will only affect received audio, i.e. on your side, and since that is fine, you probably don't need to adjust it. Instead try this: 1) Change packet size in increments of 20 ms (i.e. 0.02, 0.04 or perhaps 0.06). Your wireless link may not like too many small packets. 2) Turn off silence suppression if it's on. 3) Try a different codec -- g726-32 or even ulaw to see if it makes a difference. See if that helps. --Luki ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Issues with a Linksys SPA 2102 and asterisk
Topology: analog_phone-SPA2102-Navini_Wireless_Router--ISP--Asterisk A ping against the asterisk server shows aprox 145ms roundtrip. 128kbps upstream 512kbps downstream g729a as codec signal quality of the navini router: 100% The ATA operates correctly in every form, however sometimes when someone is talking to me (the other person is at pstn) and then I start talking the other end receives garbled voice and i need to start talking again. So I played with the jitter buffers in the available modes (low, medium, high) (direction upward, downward both) and it seems i cannot improve my voice experience. Any gurus out there with experience with the SPA 2102 against asterisk 1.2.14? thanks, -- Erick Perez ___ --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] asterisk and multiple cpus/cores
I have found a site that list the following (no date in the post, so it may be old): since all transcoding and calls still go through one core in asterisk, it doesn't make sense to buy a multi-core or hyperthreaded system that will only slow you down Does that still applies in asterisk 1.2.14/1.4.x ? Or do we have to tweak source code to balance loads (transcoding,etc) between cores? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
Indeed. The problem was the ).
thanks to all who helped me debug this...my eyes are not so young anymore...
On 2/3/07, jacobso1 [EMAIL PROTECTED] wrote:
hi,
i think the problem is here :
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
exten = _321[0123],n,Dial(SIP/${EXTEN},30,to)
note, i removed the parenthesis ')' after the {EXTEN}
this should do
regards,
jacobson
---
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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[asterisk-users] WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1'
As everybody must be watching the superbowl. I post this to let you have some fun while thinking what this can be. TDM400p (fxo) connected via loopstart to ports in an AvayaG3 call comes in from the avaya to the tdm card: WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' but call can be processed normally. comments? -- Erick Perez ___ --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] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
The following strange conditions is happening while I try to dial a
SIP user from another SIp user.
SIP to Zap dialing is fine, as all 4 users can call PSTN.
I'm using Asterisk SVN-branch-1.2-r51359M
Example: extension 3210 calls extension 3213. They are all registered properly:
chrom01*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
3213/3213 192.168.0.112D 5060 Unmonitored
3212/3212 192.168.0.112D 5060 Unmonitored
3211/3211 192.168.0.112D 5060 Unmonitored
3210/3210 192.168.0.112D 5060 Unmonitored
4 sip peers [4 online , 0 offline]
-- Executing Ringing(SIP/3210-084eaa80, ) in new stack
-- Executing AGI(SIP/3210-084eaa80,
agi://127.0.0.1:4577/call_log) in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial(SIP/3210-084eaa80, SIP/3213)|30|to) in new stack
Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213)
Feb 3 12:42:25 NOTICE[10368]: app_dial.c:1056 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
**sip.conf***
**
i have 4 extensions, 3210,3211,3212 and 3213. they are all defined in
sip.conf with the following parameters (just change 3212 for the next
extension and so on).
[3212]
username=3212
secret=3212
type=friend
context=default
nat=no
canreinvite=no
[EMAIL PROTECTED]
disallow=all
allow=ulaw
host=dynamic
language=en
dtmfmode=inband
My dial plan is like this:
The AGI is doing nothing more than simple call logging to MySQL
**extensions.conf**
**
exten = _321[0123],1,Ringing
exten = _321[0123],n,AGI(agi://127.0.0.1:4577/call_log)
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
exten = _321[0123],n,Voicemail,u${EXTEN}
exten = _321[0123],n,Hangup
comments?
--
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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[asterisk-users] asterisk 1.2 branch revision 53132 failed to compile
type codec_zap.c:389: error: dereferencing pointer to incomplete type codec_zap.c:395: error: dereferencing pointer to incomplete type codec_zap.c:396: error: dereferencing pointer to incomplete type codec_zap.c:397: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:399: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g723toulaw': codec_zap.c:415: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:437: error: dereferencing pointer to incomplete type codec_zap.c:444: error: dereferencing pointer to incomplete type codec_zap.c:444: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:445: error: dereferencing pointer to incomplete type codec_zap.c:446: error: dereferencing pointer to incomplete type codec_zap.c:452: error: dereferencing pointer to incomplete type codec_zap.c:453: error: dereferencing pointer to incomplete type codec_zap.c:454: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:456: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_alawtog729': codec_zap.c:472: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:494: error: dereferencing pointer to incomplete type codec_zap.c:501: error: dereferencing pointer to incomplete type codec_zap.c:501: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:502: error: dereferencing pointer to incomplete type codec_zap.c:503: error: dereferencing pointer to incomplete type codec_zap.c:509: error: dereferencing pointer to incomplete type codec_zap.c:510: error: dereferencing pointer to incomplete type codec_zap.c:511: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:513: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_ulawtog729': codec_zap.c:529: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:551: error: dereferencing pointer to incomplete type codec_zap.c:558: error: dereferencing pointer to incomplete type codec_zap.c:558: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:559: error: dereferencing pointer to incomplete type codec_zap.c:560: error: dereferencing pointer to incomplete type codec_zap.c:566: error: dereferencing pointer to incomplete type codec_zap.c:567: error: dereferencing pointer to incomplete type codec_zap.c:568: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:570: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g729toalaw': codec_zap.c:586: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:608: error: dereferencing pointer to incomplete type codec_zap.c:615: error: dereferencing pointer to incomplete type codec_zap.c:615: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:616: error: dereferencing pointer to incomplete type codec_zap.c:617: error: dereferencing pointer to incomplete type codec_zap.c:623: error: dereferencing pointer to incomplete type codec_zap.c:624: error: dereferencing pointer to incomplete type codec_zap.c:625: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:627: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g729toulaw': codec_zap.c:643: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:665: error: dereferencing pointer to incomplete type codec_zap.c:672: error: dereferencing pointer to incomplete type codec_zap.c:672: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:673: error: dereferencing pointer to incomplete type codec_zap.c:674: error: dereferencing pointer to incomplete type codec_zap.c:680: error: dereferencing pointer to incomplete type codec_zap.c:681: error: dereferencing pointer to incomplete type codec_zap.c:682: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:684: error: dereferencing pointer to incomplete type codec_zap.c: In function `find_transcoders': codec_zap.c:849: error: variable `info' has initializer but incomplete type codec_zap.c:849: warning: excess elements in struct initializer codec_zap.c:849: warning: (near initialization for `info') codec_zap.c:849: error: storage size of 'info' isn't known codec_zap.c:854: error: `ZT_TCOP_GETINFO' undeclared (first use in this function) codec_zap.c:859: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:849: warning: unused variable `info' make[1]: *** [codec_zap.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2/codecs' make: *** [subdirs] Error 1 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780
[asterisk-users] Re: asterisk 1.2 branch revision 53132 failed to compile
same with branch revision 53142 On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote: while compiling svn 53132 of asterisk branch 1.2 gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DZAPTEL_OPTIMIZATIONS -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o app_sms.o app_sms.c gcc -shared -Xlinker -x -o app_sms.so app_sms.o make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2/apps' make[1]: Entering directory `/usr/src/asterisk/asterisk-1.2/codecs' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i586 -DZAPTEL_OPTIMIZATIONS -DBUSYDETECT_MARTIN -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c codec_zap.c: In function `zap_framein': codec_zap.c:147: error: dereferencing pointer to incomplete type codec_zap.c:149: error: dereferencing pointer to incomplete type codec_zap.c:151: error: dereferencing pointer to incomplete type codec_zap.c:151: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:156: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:158: error: dereferencing pointer to incomplete type codec_zap.c:159: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:162: error: dereferencing pointer to incomplete type codec_zap.c:163: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_frameout': codec_zap.c:187: error: dereferencing pointer to incomplete type codec_zap.c:196: error: dereferencing pointer to incomplete type codec_zap.c:197: error: dereferencing pointer to incomplete type codec_zap.c:198: error: dereferencing pointer to incomplete type codec_zap.c:198: error: dereferencing pointer to incomplete type codec_zap.c:199: error: dereferencing pointer to incomplete type codec_zap.c:200: error: dereferencing pointer to incomplete type codec_zap.c:203: error: dereferencing pointer to incomplete type codec_zap.c:206: error: dereferencing pointer to incomplete type codec_zap.c:207: error: dereferencing pointer to incomplete type codec_zap.c:208: error: `ZT_TCOP_TRANSCODE' undeclared (first use in this function) codec_zap.c:208: error: (Each undeclared identifier is reported only once codec_zap.c:208: error: for each function it appears in.) codec_zap.c:209: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c: In function `zap_destroy': codec_zap.c:223: error: `ZT_TCOP_RELEASE' undeclared (first use in this function) codec_zap.c:224: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:227: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_alawtog723': codec_zap.c:244: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:266: error: dereferencing pointer to incomplete type codec_zap.c:273: error: dereferencing pointer to incomplete type codec_zap.c:273: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:274: error: dereferencing pointer to incomplete type codec_zap.c:275: error: dereferencing pointer to incomplete type codec_zap.c:281: error: dereferencing pointer to incomplete type codec_zap.c:282: error: dereferencing pointer to incomplete type codec_zap.c:283: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:285: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_ulawtog723': codec_zap.c:301: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:323: error: dereferencing pointer to incomplete type codec_zap.c:330: error: dereferencing pointer to incomplete type codec_zap.c:330: error: `ZT_TRANSCODE_MAGIC' undeclared (first use in this function) codec_zap.c:331: error: dereferencing pointer to incomplete type codec_zap.c:332: error: dereferencing pointer to incomplete type codec_zap.c:338: error: dereferencing pointer to incomplete type codec_zap.c:339: error: dereferencing pointer to incomplete type codec_zap.c:340: error: `ZT_TRANSCODE_OP' undeclared (first use in this function) codec_zap.c:342: error: dereferencing pointer to incomplete type codec_zap.c: In function `zap_new_g723toalaw': codec_zap.c:358: error: `ZT_TCOP_ALLOCATE' undeclared (first use in this function) codec_zap.c:380: error: dereferencing pointer to incomplete type codec_zap.c:387: error: dereferencing pointer to incomplete type codec_zap.c:387: error: `ZT_TRANSCODE_MAGIC' undeclared (first
Re: [asterisk-users] Re: Via EPIA channel_find_locked: Avoided initial deadlock
Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
you got that while doing SIP/ZAP and parking? On 1/29/07, Gordon Henderson [EMAIL PROTECTED] wrote: On Mon, 29 Jan 2007, Steve Davies wrote: I failed to notice that it was included in 51363 - I just checked, and that change is indeed already in. Sorry, my mistake. I generally do not change the -march setting, so I am probably using an i386 default. I get segfaults with the VIA C3 and C7 chips (on CN1000 and other EPIA boards) with I leave it as the defaults. I need the -i586 option. -i686 seems the be the default in the makefile. I understand it's to do with the MMX instructions used in some of the codecs... Gordon Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: Hmm. Mantis says that in SVN 51223 it was implemented, im running 51363. However I may be wrong. I will apply that patch and let you know. Thanks for the pointer. should I leave asterisk as -march=i586? or 386? On 1/29/07, Steve Davies [EMAIL PROTECTED] wrote: I would be interested to know whether this http://bugs.digium.com/view.php?id=8376 patch makes any difference. The problem is almost certainly not caused by Centos (which is widely used with Asterisk) or EPIA (which I use lots). Regards, Steve On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] detecting avaya busy tone
n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] detecting avaya busy tone
This is a G3. And I'm not the avaya operator. What do you mean with 2500 set and CPC? On 1/29/07, C F [EMAIL PROTECTED] wrote: What avaya system is this, if the avaya is configured on the ports to use a 2500 set, then it should do CPC and should work as is. On 1/29/07, Erick Perez [EMAIL PROTECTED] wrote: n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] ATCOM AT 468 manuals and firmware anyone?
both not available. but thanks. On 1/28/07, Leif Neland [EMAIL PROTECTED] wrote: Erick Perez wrote: Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I suppose you mean the AG 468 If you can find somebody who still uses Internet Explorer, the links works. The download page used to have a link for a page which worked in Firefox, but not anymore. But anyway, here are the links. http://atcom.com.cn/en/down/userguide/EN/AG-468/AG468_User_ManualGuide.rar http://atcom.com.cn/en/down/program/en/ng_series/ag468_060119_vr41rls.zip Leif ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: Via EPIA channel_find_locked: Avoided initial deadlock
I have tried compiling asterisk with -march 586 and 386 and the deadlocks minimizedin 386 but did not dissapear. Is this because of asterisk, my epia or centos? On 1/27/07, Erick Perez [EMAIL PROTECTED] wrote: In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] VIA EPIA DeadLock Issues
Via EPIA CN1 as well. Di you find any solutions? On 1/10/07, Raymond McKay [EMAIL PROTECTED] wrote: Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code) Asterisk Versions 1.2.12 - 1.2.14 - with and without metermaid patch Problem seems to happen more on systems that use parking lots. The system will run for around 24 hours or so fine, and then mysteriously, without any errors leading up to it, will stop being able to send calls to the chan_sip. System from that point on reports the following in the logs. Dec 13 12:07:04 DEBUG[16415] chan_zap.c: Took Zap/1-1 off hook Dec 13 12:07:04 VERBOSE[16415] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 DEBUG[2049] channel.c: Avoiding initial deadlock for 'SIP/100-09883f80' Dec 13 12:07:04 WARNING[2049] channel.c: Avoided initial deadlock for '0x9896848', 10 retries! attempting to stop asterisk from the CLI causes the CLI to become unresponsive and a trace shows chan_sip goes into a mutex_wait state. Anybody seen this? Have a fix? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Via EPIA channel_find_locked: Avoided initial deadlock
In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN1 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10 retries! I use stock Centos 4.4 with kernel 2.6.9-42.0.3.EL i686. I guess also asterisk was compiled for i686. and the machine is completely unusable, I need to reboot. I posted the digium script output from autosupport. It is available at: http://pastebin.com/868590 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] ATCOM AT 468 manuals and firmware anyone?
Hi there, im looking for another place that provides manuals and firmware updates for the ATCOM AT 468 and their configuration with asterisk. the site www.atcom.com.cn has non functional download links. I have several of these units but it came only with one CD, I misplaced it and I cant remember how to factory reset them and what will be the default password in the GUI. thanks for your help. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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] connecting a FXS-to-sip 4 port device to an avaya system
Thanks Jerry. Are the avaya station ports a special type ? On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote: Connect to the avaya line ports, not station ports. On Jan 18, 2007, at 10:46 AM, Erick Perez wrote: Hi, this is a signalling question: I have a 4port fxs-to-sip where i connect standard analog phones. I want to connect this device to an avaya PBX and then the device talks to asterisk via SIP. What signalling do i need the avaya to provide? FXO signalling right, like this? avayaanalog_lines_fxo_signal--(FXS side)FXStoSIPdevice(SIP side)--Asterisk thanks, -- Erick Perez ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Dimensioning a 50 sip phone installation
The customer found a VIA EPIA 1.2ghz with 1gb ram, one pci slot mini.itx. Can this equipment handle a sangoma/digium E1 card with 25 SIP ulaw phones +voicemail and *no* call recording? On 1/5/07, Leo Ann Boon [EMAIL PROTECTED] wrote: Erick Perez wrote: what if I go with full g711-no transcoding? remember that I will have an E1 coming in, so my usage can be up to 30 channels at once. if that is an overkill machine config, and for obvious reasons I cant use old hardware, what are your suggestions? I would suggest you go for a box that has redundant PSU. Most 1U boxes can't support redundant PSUs. IMHO, a 2U industrial PC with a single dual-core Pentium Dxxx 2.8GHz+ (or Xeon 3xxx) with hotswap RAID-1 HDD and PSU would be more than enough. I generally prefer 2U over 1U, because it's easier to cool and there's space to accommodate PCI cards of various sizes. Leo ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] how to transfer calls when analog phone has no transfer button
When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] how to transfer calls when analog phone has notransfer button
Don, I suppose that in order for this to work i need canreinvite=no, right? On 1/5/07, Don Pobanz [EMAIL PROTECTED] wrote: When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? Check out features.conf You can specify key presses for things such as transfer. Don Pobanz ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Dimensioning a 50 sip phone installation
what if I go with full g711-no transcoding? remember that I will have an E1 coming in, so my usage can be up to 30 channels at once. if that is an overkill machine config, and for obvious reasons I cant use old hardware, what are your suggestions? thanks, On 1/5/07, Luki [EMAIL PROTECTED] wrote: I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. Should be plenty if not an overkill. One of our setups: 20 phones, 8 outgoing/incoming SIP trunks, MeetMe conferencing with ztdummy and no Zap hardware. IVR/voice mail/MOH/Recordings/etc. Runs on a single PIII-600, 256 MB RAM. CentOS 4.4 with a stock 2.6.9-42 kernel. Asterisk 1.2.5, in production for 1.5+ years. CPU usage about 2% per call. Quite reliable (hence not upgraded). This is a g711 only setup with no transcoding. --Luki ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] how to transfer calls when analog phone hasnotransfer button
On 1/5/07, Doug Crompton [EMAIL PROTECTED] wrote: Well it would be interesting to know what FXS device you are using to connect the analog phones. I use an SPA-3000 fxo/fxs and with it you could bypass Asterisk and connect the FXO to FXS or dial directly if it were so configured, so reinvite would work but wwould probably not be desired but that is not the question... Right, I forgot to mention that. Plain an simple analog phones will be connected to audiocodes fxs-to-sip and then the audiocodes talk to asterisk. im planning *not* to use transcoding and go full g711 ulaw on this one. I am using the SPA-3000 as both an FXO (connection to telco) and FXS (connection to my house analog phones) with Asterisk in between. I have said this before on here but I will say it again. With the SPA-3000 you cannot have analog phone feature keys, transfer etc. AND still be able to use DTMF for control outside of the dialplan. If you want feature key control then you would use rfc2833 DTMF, if you want to be able to use DTMF incoming or outgoing for control then you must use inband DTMF. It is either/or. My choice was to use inband and not have features selected for the analog phones. To often I would use these phines with banking or on incoming to control voicemail functions so I wanted that capability. In that case a hook flash works fine. If you have never done it just flash the hook for a second (or use the flash key on the phone) and you will get another dialtone. Then you can call another party, tell them you have a call to transfer and hangup or click again and bring them in as a conference. Doug On Fri, 5 Jan 2007, Don Pobanz wrote: Erick Perez Don, I suppose that in order for this to work i need canreinvite=no, right? No! It doesn't matter what you have for 'canreinvite' since 'canreinvite' is a SIP attribute, not an analog phone attribute. For analog phones, Asterisk will always be in the call path. :-) -- Don Pobanz ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] no unicall on 1.4 (was: OnHook Call Announcement...)
Question: So for people using E1 with R2 or PRI as signaling, what are my options in asterisk 1.4 and 1.2? On 1/4/07, Anton Krall [EMAIL PROTECTED] wrote: Well Moises, if you do, please drop me a line and I will gladly test it. I was mentioning digium because AFAIK, the guys at digium are in touch with the programmers and contributors so I thought maybe they would have an insight on whats going to happen with unicall on 1.4, I mean, somebody at the source should know right? Many people still use unicall so I thought somebody would pick up the ball, maybe that's going to be you hopefuly. Let me know how it goes. |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Moises Silva |Sent: Wednesday, January 03, 2007 5:22 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [asterisk-users] no unicall on 1.4 (was: OnHook Call Announcement...) | |On 1/3/07, Anton Krall [EMAIL PROTECTED] wrote: | And probably wont be as Steve Underwood explained to me that he is now supporting |openpbx and has stopped support for unicall on asterisk 1.4 | | Can anybody at digium confirm? Is unicall going to be left out of 1.4? | |This has nothing to do with Digium, it has to do with anybody wanting |to code the version for 1.4, AFAIK Steve never worked for Digium and |Digium never distributed Unicall driver. | |Porting Unicall to 1.4 is in my TODO since 1 month ago, may be this |month I will have the time to give a look at the code and try to make |it work on 1.4, if somebody else cant do it before. | |Regards. | |-- |Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; |___ |--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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Dimensioning a 50 sip phone installation
Hi, Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one fonebridge (TDMoE) I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. Also, does a fonebridge setup suffers from the fact that 1.4 has no PRI/R2 support (as said in a previous post by someone else). -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Dimensioning a 50 sip phone installation
On 1/4/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Erick - Some help with dimensioning the server will be gladly accepted. -50 sip phones (g729) or g711(to avoid transcoding) in LAN -an asterisk server (1.4) doing normal pbx functions + voicemail in the same LAN -Some sporadic conferencing with no more than 2 sip phones and maybe 2 or 3 calls coming from the E1 for a total of 5 people in a conference. The asterisk server will get an E1(pri) via one fonebridge (TDMoE) I was thinking of an HP DL140 with two 250gig sata disks and one 3.8Xeon CPU with 2gig RAM. You'll do absolutely fine with this setup. I have an office that has about the same amount of phones and traffic as this (all using g711), but probably with quite a bit more conferencing. It runs on a Xeon 2.8ghz, 1GB Ram, 2 73 GB SCSI Raid 1. Also, does a fonebridge setup suffers from the fact that 1.4 has no PRI/R2 support (as said in a previous post by someone else). I've been considering buying one of these, but don't have one yet. If anyone can comment, I'd like an answer, too. - Noah ___ --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 thanks Noah, you have been very helpful. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] OT: Admin manual for Linksys Sipura SPA-2102
Hi, Anyone knows where to get the admin (not the end user) manual for the linksys spa2102. This model is the 2 analog port+router. There are a lot of advanced options that I would like to see what they do. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] DB9 e1 to RJ45 pinout
from my aging memory. Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim DB9 and make standard RJ45 jack. Also, http://www.pccables.com/01910.htm http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 On 11/24/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] DB9 e1 to RJ45 pinout
I forgot to mention that in the db9 part I guess pins 2-3(tx) and 6-8(rx) are used. I'm sorry I dont recall ground. On 11/24/06, Erick Perez [EMAIL PROTECTED] wrote: from my aging memory. Standard E1 pinouts for RJ45 jack are 1-2 (TX) and 4-5(RX). So, trim DB9 and make standard RJ45 jack. Also, http://www.pccables.com/01910.htm http://www.zytrax.com/tech/layer_1/cables/tech_rs232.htm#t1 On 11/24/06, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all, anyone known how can i create an adapter from a DB9 port of a E1 to an RJ45 plug? My telco left active the db9 port, but on my te407p card i have rj45 connection. Anyone can help me pls ? Thanks in advance -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.430 / Virus Database: 268.14.14/548 - Release Date: 23/11/2006 15.22 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk voicemail and hotel software integration
Good Evening, does anyone have information regarding integration of asterisk voicemail with an hotel management software called Fidelio made by the Micros Company. The integration can be either opensource or paid. please contact me offlist if you want. Thanks, Erick. eaperezh (at) gmail (dot) 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
[asterisk-users] make: execvp: build_tools/make_svn_branch_name: Permission denied
Hi, I got a svn trunk of zaptel, while doing make clean or make (as root) I get: make: execvp: build_tools/make_svn_branch_name: Permission denied I am not sure what the error can be. thanks for your help. -- Erick Perez ___ --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] DTMF Corruption Problem
I can report that with asterisk 1.2.13, internal SIP calls work perfectly but (in my particular case) my asterisk box cannot recognize DTMF digits when it receives a call via our SIP provider. we are both using rfc2833 and I have tried relaxdtmf=yes/no when i use an internal sip extension and call somebody outside via my sip provider, dtmf is recognized. On 11/9/06, mail-lists [EMAIL PROTECTED] wrote: Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? I really don't know for certain but here's what I experienced: When calling out asterisk gives the option to allow called numbers to transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th dial string. This would very seldom work. I could hit the '#' on the called phone it would say 'extension' but would always reply with 'not valid extension' I recently upgraded to 1.2.12 and noticed that there was no ztdummy running! I compiled my own zaptel installed it, loaded the modules on boot and now the transfer works perfectly. Also: my moh wasn't working for some reason. After I installed the ztdummy module it works too.. I'm not sure whether the transfer issue was fixed by using the ztdummy module or by the asterisk issue but my point is that you should always have the ztdummy module installed if possible. Just my .02. Hope it helps ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] DTMF Corruption Problem
On 11/9/06, mail-lists [EMAIL PROTECTED] wrote: Erick Perez wrote: I can report that with asterisk 1.2.13, internal SIP calls work perfectly but (in my particular case) my asterisk box cannot recognize DTMF digits when it receives a call via our SIP provider. we are both using rfc2833 and I have tried relaxdtmf=yes/no when i use an internal sip extension and call somebody outside via my sip provider, dtmf is recognized. On 11/9/06, mail-lists [EMAIL PROTECTED] wrote: Also, I am not using a zaptel timer. Could this possibly be causing problems with DTMF?? I really don't know for certain but here's what I experienced: When calling out asterisk gives the option to allow called numbers to transfer by hitting the '#' by putting 'T' (or 't'?) as an option in th dial string. This would very seldom work. I could hit the '#' on the called phone it would say 'extension' but would always reply with 'not valid extension' I recently upgraded to 1.2.12 and noticed that there was no ztdummy running! I compiled my own zaptel installed it, loaded the modules on boot and now the transfer works perfectly. Also: my moh wasn't working for some reason. After I installed the ztdummy module it works too.. I'm not sure whether the transfer issue was fixed by using the ztdummy module or by the asterisk issue but my point is that you should always have the ztdummy module installed if possible. Just my .02. Hope it helps ___ --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 Erick, Do you have ztdummy running? What SIP provider are you using. Incoming calls work fine for me (and always have as far as I know). ___ --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 I have a TDM400 installed. loading wctdm and not ztdummy. centos 4.4 with kernel 2.6 My provider is not located in US...I am not lcated in the US -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] names of SIP aware firewalls
Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] compilation problem with asterisk-addons
Thanks, I learned the hard (but fun) way!!! Cheers, On 11/1/06, Russell Bryant [EMAIL PROTECTED] wrote: Erick Perez wrote: Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make clean make and the output is: ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory You need to install Asterisk before trying to compile and install Asterisk-addons. -- Russell Bryant Software Engineer Digium, Inc. ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] VOIP Bandwidth questions
this it seems that even calls to other large voip providers go through the PSTN though). Barring voip to voip calls, everything must run through their bandwidth right? If I'm right on this, I guess we need to come up with some sort of viable business model to do sell our own service. I want to concentrate on smb clients to whom we can then provide an asterisk box which would leave our bandwidth free, but my boss isn't particularly keen on this route. Anyways, Thanks for any insight and advice on this question, sorry if I'm asking this in the wrong place Thanks, Steve ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] VOIP Bandwidth questions
I forgot to tell that my rant is about a centrally handled servers, with no re-invite and no spider-like interconnects with smaller, geographically located switches. On 11/2/06, Erick Perez [EMAIL PROTECTED] wrote: This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. For example: Telco A has 100 subscribers to his phone service in a city (home and business), so he needs to ask himself a- Will the telco buy a switch that can handle 100 calls simultaneously? So he can provide service to his subscribers 100% of the time at any time of the day even during riots,panic,flood,etc? b- Or will the telco go for a balance and guess that at the peak time of the day he will have 75 simultaneous call, so he goes out and buy a switch that handles 75-80 calls at the same time? c- how many trunks will the Telco have to talk to other telcos? So telco in City A can communicate with Telco in city B (or even in the same city)? International voice providers suffer from this kind of problem. Some sell plastic cards with a local phone number and a pin so you call them to call to other cities/countries but that cheap voice provider has, let's say, ten thousand long distance lines and ten thousand local phone numbers, but they sell 100k plastic cards a month with a peak usage 3 times every ten days of 12thousand lines? obviously 2 thousand callers wont get connected (only 3 times every ten days in a specific time range) but the other 7 days the peak usage is 10thousand calls? Every ten days the provider try to connect 106k calls but fail to connect 6k calls, that's 6% failure rate every ten days (100% in a 7 days period and 98% in those 3 days). Can you live with that failure ratio? that's up to you. I don't work for a Telco, but a Telco may apply the dialup-internet rule (and they live happy with it) for 30subscribers-to-1line home users and 10(or 5)subscribers-to-1line for business. (correct me if I'm wrong please it will be nice to know real figures). So apply the same rule to you VoIP hosting. -What codec will you use? let say g711 and let's say it uses 100kilobits per leg. -How many subscribers will you have in a 6 month period? 500 -So to provide all of them with service you will need 48Megabits of bandwith all the time just to connect to your Telco equipments. - But you decide that you analyzed the usage patterns of your service and you will have only 125 subscribers calling other 125 subscribers (this is called On-Net) at peak time every day at 6pm (rush hour). So, go out and buy 24mbits of bandwidth only. - But you suddenly have the option to hire burst IP service where your IP carrier can provide you with more bandwidth if your usage starts to rise in any given time of the day. So you calculate again that your minimum constant usage at any time of the day is 40 users On-Net, so go out and buy 5mbits (for a total of 50 calls) of bandwidth with burst IP enabled from 6pm to 8pm of 48mbits (or 24mbits). This scenario is only subscriberyour_companysubscriber. you also need to calculate subscriber--your_companyother_telcos And the last but most important question is: how much money do you have to burn on this? 100% Uptime full-service, Top Carrier Class performance (and even they get busy sometimes)? or almost perfect service with the once-in-awhile glitch of we're sorry all circuits are busy, please try again. On 11/2/06, mail-lists [EMAIL PROTECTED] wrote: Hello everyone, This probably isn't the correct place to ask this but I thought I'd check here first. We're getting ready to roll out a hosted pbx solution on a very limited trial basis (some company employees are going to get voip service at home). Our main issue is of course bandwidth. We have enough bandwidth (spread across two locations) to accommodate the few employees (around 10) for the near future but we're worried about how this is going to scale. Obviously at some point we'll need to consider 'real' bandwidth. My question is this: How do huge voip companies like vonage handle bandwidth. I'm pretty sure that they have to have sufficient bandwidth available for X numbers of simultaneous calls, in other words ALL VOIP traffic runs through their servers, right? My boss is of the mind that there is no way that this is a viable business model and his insistence has me doubting myself. So, to clarify - Vonage has to have the necessary bandwidth to handle whatever amount of simultaneous calls. I can imagine that one vonage user calling another vonage user would use some sort of sip re-invite and perhaps even calls to other huge providers (packet8) are direct client to client. (Last time I read about this it seems that even calls to other large voip providers go through
Re: [asterisk-users] Error installing asterisk, module zaptel not found
is zaptel.ko anywhere in your system? it should be in /lib/modules/`uname -r`/extra/ On 11/2/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: After deciding to move a semi working asterisk setup to another box, installing and recompiling asterisk, addons and zaptel, modprobe zaptel says, module not found. Following various tales of how to modify udev stuff, still get that error. lspci does show the board in the list. All the LED's on the back of the board are dark. I have a TDM400p (tdm22b). I did not actually install the board, until after asterisk and add ons were complied. Just before the steps to compile zaptel. After installing board and playing doing the udev hack dance, did recompile with same results, as stated. What could be the probem(s)? phoneman ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] compilation problem with asterisk-addons
Hi, Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this: Note: MySQL libraries are installed and the structure is as follows: /usr/src/astsources/asterisk-1.2.13 /usr/src/astsources/asterisk-addons-1.2.5 in /usr/src/astsources/asterisk-addons-1.2.5 I do: make clean make and the output is: ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory app_addon_sql_mysql.c:30:34: asterisk/linkedlists.h: No such file or directory app_addon_sql_mysql.c:31:31: asterisk/chanvars.h: No such file or directory app_addon_sql_mysql.c:32:27: asterisk/lock.h: No such file or directory app_saycountpl.c:11:27: asterisk/file.h: No such file or directory app_saycountpl.c:12:29: asterisk/logger.h: No such file or directory app_saycountpl.c:13:30: asterisk/channel.h: No such file or directory app_saycountpl.c:14:26: asterisk/pbx.h: No such file or directory app_saycountpl.c:15:29: asterisk/module.h: No such file or directory app_saycountpl.c:16:27: asterisk/lock.h: No such file or directory cdr_addon_mysql.c:23:29: asterisk/config.h: No such file or directory cdr_addon_mysql.c:24:30: asterisk/options.h: No such file or directory cdr_addon_mysql.c:25:30: asterisk/channel.h: No such file or directory cdr_addon_mysql.c:26:26: asterisk/cdr.h: No such file or directory cdr_addon_mysql.c:27:29: asterisk/module.h: No such file or directory cdr_addon_mysql.c:28:29: asterisk/logger.h: No such file or directory cdr_addon_mysql.c:29:26: asterisk/cli.h: No such file or directory res_config_mysql.c:41:30: asterisk/channel.h: No such file or directory res_config_mysql.c:42:29: asterisk/logger.h: No such file or directory res_config_mysql.c:43:29: asterisk/config.h: No such file or directory res_config_mysql.c:44:29: asterisk/module.h: No such file or directory res_config_mysql.c:45:27: asterisk/lock.h: No such file or directory res_config_mysql.c:46:30: asterisk/options.h: No such file or directory res_config_mysql.c:47:26: asterisk/cli.h: No such file or directory res_config_mysql.c:48:28: asterisk/utils.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/astsources/asterisk-addons-1.2.5/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o common.o common.c common.c:1:29: asterisk/logger.h: No such file or directory common.c: In function `decode_header': common.c:93: warning: implicit declaration of function `ast_log' common.c:93: error: `LOG_WARNING' undeclared (first use in this function) common.c:93: error: (Each undeclared identifier is reported only once common.c:93: error: for each function it appears in.) make[1]: *** [common.o] Error 1 make[1]: Leaving directory `/usr/src/astsources/asterisk-addons-1.2.5/format_mp3' make: *** [format_mp3/format_mp3.so] Error 2 Thanks for your help. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
I forgot to mention that the Carrier that owns the ATA box was not willing to let me connect directly over IP, I was only allowed to use the FXS port. He already ack that he has a problem with disconnections. On 10/31/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote: Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 What exactly is the point is such settings? Why not connect directly to the provider over SIP? Or to the ATA over SIP? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme server for 8 more calls. Everything works fine in terms of the asterisk/meetme. The issue arises when the calls comes in via the ATA286 box and in any part of the meeting the CALLER hangs up but the ata286 does not realize the caller hung up so the channels remains open and everyone in the room hears a busy signal. After 30 seconds the ATA286 hangs up (I guess due to timeout) and then the tdm04b hungs the channel and then the meetme room gets back to normal. This is an ATA286 issue right? nothing to do with the TDM or the asterisk box? Since I do not own the ATA286 (the voip provider does) would you recommend something to be asked/changed to the provider of the ATA? Thanks, -- Erick Perez ___ --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] Re: 0 channels configured with tdm400 (tdm04b rev. G)
More info. [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. cat /etc/modprobe.conf alias scsi_hostadapter ahci alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd alias eth0 e100 alias eth1 3c59x alias wcfxs wctdm install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg *** [EMAIL PROTECTED] ~]# cat /etc/asterisk/zaptel.conf fxsls=1-4 loadzone=us defaultzone=us *** [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 [EMAIL PROTECTED] ~]# On 10/27/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Re: 0 channels configured with tdm400 (tdm04b rev. G)
Tzafrir, please disregard my previous postdefinitely i was WAAAY sleepy. The error was simple (now that I just waked up completely): I was touching /etc/asterisk/zaptel.conf instead of /etc/zaptel.conf. I should remind myself not to work past midnight. I don't recall what made me think the file was inside the asterisk directory. Thanks and apologies to all,. On 10/28/06, Erick Perez [EMAIL PROTECTED] wrote: More info. [EMAIL PROTECTED] ~]# ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. cat /etc/modprobe.conf alias scsi_hostadapter ahci alias usb-controller ehci-hcd alias usb-controller1 uhci-hcd alias eth0 e100 alias eth1 3c59x alias wcfxs wctdm install wctdm /sbin/modprobe --ignore-install wctdm /sbin/ztcfg *** [EMAIL PROTECTED] ~]# cat /etc/asterisk/zaptel.conf fxsls=1-4 loadzone=us defaultzone=us *** [EMAIL PROTECTED] ~]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 1 WCTDM/0/0 2 WCTDM/0/1 3 WCTDM/0/2 4 WCTDM/0/3 [EMAIL PROTECTED] ~]# On 10/27/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Is SQLite a replacement for Mysql while using ARA in 1.2.x
Cohen, so you vote for the ARA-odbc-sqlite route? this is for embedded, so that's why sqlite instead of mysql or postgres. when you say it is not guaranteed, what do you mean? On 10/27/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Oct 27, 2006 at 07:56:16AM +0800, Leo Ann Boon wrote: Moises Silva wrote: AFAIK, you will need to do the first. ARA-odbc-sqlite res_sqlite3 in asterisk-addons supports ARA res_sqlite3 from aadd-ons is a strange beast. It uses its own, private copy of sqlite and acceses internal data structures. So while the database that it uses is hopefully sqlite3, it is not perfectly guaranteed. (This is why it's not part of the Debian packages built of asterisk-addons) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] 0 channels configured with tdm400 (tdm04b rev. G)
Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel = 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with Asterisk Version 1.2.13 Zaptel Version 1.2.10 Libpri Version 1.2.4 Physically looking at the card, the four FXO ports have the green led turned on. It has no IRQ conflicts and zaptel compiled cleanly. Kernel is 2.6.9-42ELsmp (it's a dual core Intel machine in an Intel 945G board) Your comments are welcomed. ___ --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] res_sqlite problems
Hi Michael, do you have any new information about sqlite and asterisk in realtime? what release of asterisk are you using? On 8/7/06, Michael Iedema [EMAIL PROTECTED] wrote: Greetings, I'm trying to replace my extensions.conf with a sqlite database. So far everything's gone really rocky to be honest with you. I do, however have it up and running with a few minor cli messages complaining about the missing 'h' extension, etc. Problem: I'm trying to stress test asterisk a bit to see the performance difference between the static extensions.conf and the realtime one. I've generated a 1 entry extension with Answer, many NoOp's and Hangup to do this. res_sqlite segfaults asterisk when attempting to go through this. I've messed with the res_sqlite code and noticed that the needed memory's being statically allocated beforehand. Increasing the array solves the problem but it makes me wonder about the resource's flexibility. Questions: Is anyone using res_sqlite to do really heavy lifting at their install? (multiple queries, views, etc) Can anyone vouch for it's stability in general? Does anyone have any additional documentation related to it? There seems to be a real lack of docs on this one. Thanks in advance, --Michael I. ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Is SQLite a replacement for Mysql while using ARA in 1.2.x
Can I safely assume that SQLite can be used to code something for Asterisk Realtime instead of the much used mysql database? I have read several old posts, but nothing point me to an answer. maybe ARA--odbc--sqlite or ARA--sqlite? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Electric usage of a tdm400p
Well Im planning to use a mini-itx, a laptop hdd and a 4fxs digium card. the mini-itx comes with a 60W DC to DC adapter (80W peak). So I need power to manage the hdd, motherboard,the tdm card. A disk cable can be made available, but is not present as a factory default. So My real concern is power. On 10/18/06, Bob Chiodini [EMAIL PROTECTED] wrote: On Tue, 2006-10-17 at 11:59 -0500, Erick Perez wrote: Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? Erick, Per http://en.wikipedia.org/wiki/Ring_(telephone) in the US the ring voltage is around 90VAC (20 Hz) with a current of 30 milliamperes (REN ~5). This translates to 2.7 watts. Assuming a DC/DC converter efficiency of 38% (probably low), you would need about 3.7 watts, per FXS module. About 15 watts, total. What is the TDM card installed in and is a disk drive cable available? Bob... ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Electric usage of a tdm400p
Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? -- Erick Perez ___ --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] 1.2.12.1 crashing
Maybe a total dumb question but I see you talk about the 1.0.x version and the 1.2.x version. I always see references to the 1.2.x version. Where can I read about the differences in 1.0 and 1.2? Isn't the 1.0 version only available when you buy ABE ? On 10/13/06, Joseph [EMAIL PROTECTED] wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: On Thu, 12 Oct 2006, Eric ManxPower Wieling wrote: Matt Florell wrote: If you downgrade, let us know if it fixes things for you. It's strange that there were so many changes in the 1.2 SVN branch after 1.2.7.1 that seem to be complete changes in how some things operate(like the transcoding optimization mess for Asterisk 1.2.11 and 1.2.12 that was fixed in 1.2.12.1). I wish that such radical changes would not be made in a release branch at the expense of reliabitily. Maybe Digium can run the next release for 7 days on their PRODUCTION Asterisk box before a release. I guess they did, and it probably worked. Then they run it for several months, and if it works they label it Business Edition and actually sell it because they know it will work. What hardware are they testing it with, just Digium cards? Asterisk 1.2.12.1 definitely doesn't run correctly with Sipura 3000, as it crashes on second call to PSTN line. -- #Joseph ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] OT: BioFuel to power phone networks
This are the things that make me believe in technology. I wonder if Ubuntu Linux advocates will help with the development of the controlling modules. * Reuters 16:55 PM Oct, 11, 2006 AMSTERDAM -- Palm and pumpkin seed oil could soon be generating electricity to help power cell phone networks across Africa under a plan to replace fossil fuels with sustainable biofuels made from crops grown by local farmers. Full Story: http://www.wired.com/news/wireservice/0,71936-0.html?tw=rss.technology -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Issues with Asterisk 1.4 Beta
My 2 cents but im still playing with 1.4 Issue 5: on the phones disable silence supression or set to yes the transmit silence option. I am not sure if that is the nameof the option in Swiss phones but the whole idea is to *not* save bandwidth when the line goes silent (because both sides stop talking). Make sure ALL SIP phones have disabled silence suppression you may as well take a look at: bug 5374, which allows Asterisk to communicate with devices that support silence suppresion; bug 5409, comfort noise generation in Asterisk; and bug 1234. cheers, On 10/12/06, Jason Walker [EMAIL PROTECTED] wrote: I thought I would list my issues so all of you that know more than me might be able to help. 1. I have 6 Swissphone ip10 they disconnect calls at either 70 seconds, 120 seconds or 180 seconds I have polycom Phones that go forever 2. When I try and transfer calls I have a LONG delay before the seconds line is usable. Call1 on hold then make second call and 1 minute passes before it attempts a connect 3. I have many Polycom 501s and I cannot seem to get the tick server to work. I change settings but it does nto fetch the time 4.I get-- Got SIP response 500 Internal Server Error back from 192.168.0.XXX from all my Polycom 501 phone every 2 mintues or so 5. I get [Oct 12 08:49:56] NOTICE[29165]: rtp.c:708 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.0.141 on my Swiss phones Any help would be great. I am a little new to asterisk and so if I posted this incorrectly please let me know Jason Walker ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk RT on Disk On Module Performance and Durability
Jeremy, Cohen, Kris, thanks to all of you. Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) ) the whole idea is for a customer that needs an IVR in 4 languages with autoattendant, extensive CDR and plotted usage patterns as well as voicemail. Voicemail will be used *a lot*, probably about one thousand voicemails per day and the customer does not want VM-to-Email (God knows why!). Oh, and the whole idea of the database is because the developers are working in an AJAX based interface that does the asterisk config/plotting/vm/day-to-day stuff with ARA, so a db is needed. I started learning asterisk with flat files...it works for me...but hey...times are changing. Who knows, maybe the whole thing can be fitted in ram (except for the vm part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds and extra breakup layer (maybe Im kind of outdated). Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM. Again, Thanks to all of you. P.D. I will later follow this thread with the full working configs that will take place at user premises. And for the sake of the test. I will try to kill a sandisk USB with the full config. On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after. You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper: http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf While it specifically discusses their industrial line of CF cards, it is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will noticethat the SHORTEST expected life of a CF card in their test scenarios was over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is stillseven years.I expect to get at least that from my original AstLinux system.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.Theyare meant to be used directly on flash memory and do their own wear leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is thebest FS to use.CF cards and DOMs use their own wear leveling, so none is required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions.With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cardswill outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long! :) To get back to answering your question, I HIGHLY recommend that youavoid MySQL and realtime on your box with a DOM.Nothing against either(MySQL or Realtime), but they will probably make your device more complicated than it needs to be while substantially shortening the lifeof your DOM.If you absolutely have to use MySQL, you might have betterluck using a MySQL storage engine that uses fewer writes than InnoDB, but I am no expert on that.--Kristian Kielhofner___--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 -- Erick PerezPanama
Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability
Douglas, Im just the asterisk guy. If they decide to write a cross-browser multi-tier interface in AJAX, assembly language or Pascal, that's up to them (the programmers). I will let them know what can/can't be done. Thinking of that...15 years ago...the last time i used pascal. On 10/9/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm just going to jump in here, and ask a stoopid question. How could you possibly write a multi-user front end in AJAXwithout using a database backend like MySQL? Doug. -Original Message-From: Erick Perez [mailto: [EMAIL PROTECTED]]Sent: Monday, October 09, 2006 1:58 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk RT on Disk On Module Performance andDurability Jeremy, Cohen, Kris, thanks to all of you. Indeed after reading the Sandisk paper it shed a lot of light on this matter. The whole idea is to have a large scale system with no moving parts (we call a large system something with250 users, at least down here ;-) ) the whole idea is for a customer that needs an IVR in 4 languages with autoattendant, extensive CDR and plotted usage patterns as well as voicemail. Voicemail will be used *a lot*, probably about one thousand voicemails per day and the customer does not want VM-to-Email (God knows why!). Oh, and the whole idea of the database is because the developers are working in an AJAX based interface that does the asterisk config/plotting/vm/day-to-day stuff with ARA, so a db is needed. I started learning asterisk with flat files...it works for me...but hey...times are changing. Who knows, maybe the whole thing can be fitted in ram (except for the vm part)...we'll see. I had to ask anyway, but i don't like Dbs eitherit adds and extra breakup layer (maybe Im kind of outdated). Smaller iPBXs will definitely be CF and RAM based and I, at least, will force VMtoEmail and do all the processing in RAM. Again, Thanks to all of you. P.D. I will later follow this thread with the full working configs that will take place at user premises. And for the sake of the test. I will try to kill a sandisk USB with the full config. On 10/8/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Jeremy McNamara wrote: Tzafrir Cohen wrote: H, I'm not sure that this is exactly the data you're after. You're looking for the ammounts of writes for the disk block that gets the most writes. E.g: for a standard ext3 filesystem, the journal area would probably have very frequent writes, whereas most of the system would remain mostly unchanged. Again, if the embedded system is setup properly, there is NO writing to the flash during normal operations, thus the device won't be killed by its alleged 2 million write limitation. Kris and I had a quick discussion on this topic, off-list, and his original flash-based device is still in constant operation after 2 years and I have flash modules that I purposely tried to kill with writes. It took significant effort to start causing error situations, which were very easily detected before the system would become unusable. Erick, you should focus on having a quick action restoration plan and extra DOMs always readily available.Then when a failure situation is detected, you can react very quickly. Jeremy McNamaraJeremy, Erick - I have always pointed to this SanDisk whitepaper: http://www.sandisk.com/Assets/File/OEM/WhitePapersAndBrochures/RS-MMC/WPaperWearLevelv1.0.pdf While it specifically discusses their industrial line of CF cards, it is pretty obvious that flash can, and often does, last much longer thanother components in a system when properly implemented.You will notice that the SHORTEST expected life of a CF card in their test scenarios was over 70 years!How long is your power supply going to last?Even ifthe consumer level cards had 1/10 the life expectancy, that is still seven years.I expect to get at least that from my original AstLinuxsystem.It's been two so far, I'll let you know how it is doing inanother five years :). JFFS (and similar FSs) are not appropriate for CF cards or DOMs.They are meant to be used directly on flash memory and do their own wear leveling and in some cases, compression.All kinds of commercialdevices use JFFS2.If you are using a CF or DOM with Linux, ext2 is the best FS to use.CF cards and DOMs use their own wear leveling, so noneis required in the operating system or file system.CF cards and DOMshide wear leveling from you and expose themselves as an ordinary IDE device. I echo Jeremy's conclusions.With a properly designed operating system, decent flash memory, and a reasonable usage pattern, I can tellyou (with a great amount of certainty) that in most situations, CF cards will outlast just about any hard drive (even SCSI) when used 24/7. These days, it really is pretty tough to trash flash. However, if you are running a MySQL cluster or something with several,multi-gigabyte databases, no type of flash memory will last very long
Re: [asterisk-users] Asterisk RT on Disk On Module Performance and Durability
I understand Jeremy and Kris point of view (BTW Kris, astlinux rocks!!) However the main question was not aswered (or i didn't get it, did I ?) If I use a Disk on Module that has 2million hours MTBF and a Read/Write lifecycle of 2million times, then, How many days/weeks/months/years will take to do 2million read/write cycles? which leads to my second question. How do I measure/count the read and writes a normal linux system running asterisk does during a day, so I can extrapolate that in terms of time? Is there an utility? Example: if I setup system XYZ with asterisk, then load this magical utility/procedure that counts how many writes the filesystem has done to / or to /,/tmp,/var and after 24 hours the utility/procedure says: 10thousand writes, then, I will do 10thousand writes a day multiplied by200 days= 2 millions Obviously this means I will not use a RAM disk and I want to write to the module everytime Then i will assume that the Disk on a Module will die after 200 days. Or am I completely and horribly misunderstanding the 2million Read/WriteLifeCyleadvertised by Disk-on-Module companies? Example: http://www.pqi.com.tw/product2.asp?oid=140cate1=143PROID=34 ‧MTBF:2,000,000 Hours‧R/W Cycle:2,000,000 Times I want to understand if that's what they mean. I fully understand that such media will have a longer life cycle if i only read from it and keep writes to a mimimum, for example: writing dialpan changes. The whole idea comes from doing a mini itx with no moving parts offering voicemail stored in a disk-on-module and astlinux in a CF and a RAM Disk large enough to do processing on RAM before saving to CF or to disk-on-module when needed. Thanks again for you comments, On 10/6/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kristian Kielhofner wrote: Erick, OrJust use AstLinux which kind of does what Jeremy described :) http://www.astlinux.org P.S. - I am the creator of AstLinux -- Kristian Kielhofner Sorry to reply to my own post, but there seems to have been some confusion in what I said here.To completely clear it up, Astlinux onlywrites to flash in these circumstances:1)You update the configs.2)You update AstLinux.3)You are using voicemail and people leave voicemail. (most flash seems to last long enough given typical voicemail usage patterns)4)If you have the PERSISTLOG option enabled, I will save syslogs toflash (not RAM - the default).Users are warned about this, and it is not the default.5)astdb is stored in flash, so depending on your needs, SIPregistrations and/or dundi keys may get written here periodically.Imight make an option similar to PERSISTLOG to disable this. Also, you have the option of using a hard drive or alternate flashdevice for ALL writes.Boot from flash, run from HD.Do whatever worksbest for you and your application.--Kristian Kielhofner ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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] Asterisk RT on Disk On Module Performance and Durability
Hi, Im doing some research for Disk on a Module (DOM)with asterisk realtime. To have no moving parts for a special project, I know I can use 3.5 or 2.5 HDDs but DOMs sound interesting. Does someone have working experience with this? Basically the Asterisk Realtime will be stored in MySQL and the DB will be stored in a Disk on a Module. I have read that the usual standard is 2,000,000 MTBF and 2,000,000 Read/Write Cycles. Is there an utility/section/procedure that can count/display the reads and writes a normal Linux system does? That result can be extrapolated to understand, in terms of days/week/months how much time a Disk on Module will last. Anyone with field experience? Thanks, -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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] Are you using app_meetme or app_conference
Hi, for call centers with voip phones and calls coming in via SIP and Zap, what app_ are people using to do: -conference -listening to conversation of agents Is app_meetme or app_conference? Does app_meetme still suffers from the need to transcode to slin? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Mini call center only 15 seats fxs to sip suggestion
Hi, I looking for an affordable (maybe used) FXS to SIP media gateway (or another method) to be deployed in a mini call center. The final user already has analog phones and a cabling setup in place. The cheap gateway will send and receive SIP traffic to an asterisk box that is already in place and connected to PSTN. The asterisk is there because it will provide voice recording and voicemail to email and a simple IVR. The final user does not want to spend the money associated with items like and audiocodes gateway or a sngoma remora or digium FXS card. that's why we are looking for a media gateway. Since he already have some analog panasonic phones, he does not want to purchase Ip phones. if you have some other ideas, let me know. Ebay turned nothing in my searches. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] How much SIP calls can I squeeze from this box
Hi lists, I would like to know how much can i get from the below configuration. I have a machine in my office that I want to use for demo purpose. The features I want to implement are: voicemail (users call the box to get their messages) voicemail to email (some users will the the vm by email) pbx like behavior (music on hold, a simple IVR to select what department to talk to) Full 100% call recording. Software spec: Centos 4.4 Asterisk 1.2.12.1 no sql SIP users with IP hardphones running g711 Hardware: Asterisk Box: Dual core Pentium D at 2.4ghz, 533fsb, Intel 945GNT board,100Mbit intel NIC. Dual 80gbit sata2 disk. A 8-port fxs card (pci in a PCI-X slot) and the FXS will be connected to a Panasonic PBX Protocol: G711 all the way if possible (even in moh) SIP users?: Here it comes my question in terms of: - Registered users - Simultaneous calls (remember full call recording) BTW: What options do I have to minimize disk writes for the call recording part? more ram to make it as a ramdisk? special ramdisk cards? any special format or way to capture/encode/store the recorded stream? During night hours I was thinking of moving the recorded files to another server via NFS. thanks in advance. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] OT: Bandwidth calculations and PCI/PCIX/PCIE
I found this interesting but old white paper at Dell.com tech solutions and another one from INTEL. It compares bandwidth usage of a PCI, PCI-X, PCI-E in33/66/100/133mhz bus and different technologies that can saturate the bus. It helped me understand the bandwidth required for TDM (sangoma/digium) cards and how far can I push the PCI bus in an old and newmotherboard. I hope it help others to understand how much a network card can pump and make calculations about consumptions in TDM cards. make sure the link is a one-line in your browser Original online document http://www.dell.com/content/topics/global.aspx/vectors/en/2004_pciexpress?c=uscs=08Wl=ens=bsdv here is the link to the same Dell article but in PDF form. http://www.dell.com/downloads/global/vectors/2004_pciexpress.pdf Another interesting document from INTEL www.intel.com/technology/pciexpress/devnet/docs/WhatisPCIExpress.pdf The facts learned from these documents are: a- 3.3volts/32bit PCI cards can be used in PCI-X slots. (i just discovered that, sorry forliving under a rock) b- The slowest PCI card in Mhz will dictate that PCI-X bus speed. So avoid degradation by not installing a PCI card and a PCI-X card in the same bus (check you motherboard design), your motherboard design usually have two buses. c- If you use a PCI-X based implementation motherboard, you will not saturate the bandwidth of the board, using Quad or Octal port cards (e1/t1/j1). -- Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers
Hi, I would like to read your comments for the following setup: Building A: 3 voice E1incoming toa quad redfone fonebridge (TDMoE) The fonebridge goes to a port in a 24 port gigabit switch in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server in the gigabit switch VLAN2 is for the second gigabit NIC card on the server andeleven 10/100 switches with 250 SIP phone users running g711 codec (24 phones per 10/100 switch,each switch is 24port) Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building A are 1xxx Building B: same config E1/switch/users as building A Building A and Building B are connected over a 10Mbits fiber link. Numeric Extensions at building B are 2xxx The asterisk servers at each side will talk IAX2 between each other for building-to-building call transfers. Suggested machine: Im considering a Dell PowerEdge 9G 1950, Dual Xeon 3.20Ghz, 1066 FSB, 4GB ram. two 73GB SAS 15k RPMs hard disk and dual gbit network card. Asterisk Features: Music on hold call transfer call waiting (but only on executive phones, around 20) voicemail a small queue (about 10 persons) and a simple IVR (play prompts for department selection, transfer according to selection). No call recording requested at this time. Operating System: Centos 4.3 Codecs: G711 for the SIP to asterisk and IAX for server to server transfers. If IAX is not recommended, please advice. Notes: a- Is is expected to have the 250 SIP users talking either to each other and/or to the other building and/or to the fonebridge E1s. b- I know that for SIP-to-ZAP a calculation of 30Mhz per voice channel is a rule of thumb, but i also read somewhere that the same calculation does not apply when doing Pure IP, no SIP/ZAP and pure g711 implementations I'm in that category. c-Just for the record, what if I change to g729? d- It is expected to have 80% of the calls over the E1 being incoming from the PSTN and the other 20% ar the SIP users calls to the PSTN Is is also expected to haveone 24 port Rhino FXS channel banks connected to the 4th port of the fonebridge. Is used, it will add another 24 users to the setup. Thanks in advance. Your comments are welcomed. Erick PerezPanama SistemasIntegradores de Telefonia IP y Soluciones Para Centros de DatosPanama, Republica de PanamaCel Panama. +(507) 6694-4780 ___ --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] Asterisk load testing
Nitin: Some generalized specs: A voice call takes aprox. 30MHZ of CPU. In your spec, a dual 240 (1.4Ghz) may take up to: 1400/30=46 calls x 2 = 92 calls Im just talking G711 here. I have not taken into account if you're going to use voicemail, AGI, etc,etc,. Just plain calls. I also have not taken into account how many phones can register to this machine. Personally, I make calls, not registrations, so it is useless to me to know that a billion phones can register to a given asterisk machine but only 100 can make calls. So, my personal point of view is that your machine can do 92 calls (SIP TO ZAP) at full g711 quality with at least 4 times the registrations (that means about 400 phones can register). However, due to the CPU structure of Opterons, that number may be a little high. As Martin said, look the archives. There are gallizions of configurations that can help you, or, use/rent products like ABACUS or the asterisk load tester. And about howe much internet bandwidth a codec requieres, well, look for the codec size/payload and add a few kilobits of IP overhead. Example: G711 is 64 kilobits per second, a conservative figure will be to add 16 kilobits of overhead so the total size of a g711 transmission will be (64+16) 80 Kilobits per second per leg. When you see the term per leg it means this: SIP user/g711-80kbps(first leg)-Asterisk80kbps(second leg)-destination That means each side of the conversation will take 80Kbps of bandwidth. Hope it helps, feel free to ask again and welcome to the list. Cheeers, On 8/14/06, Nitin Gupta [EMAIL PROTECTED] wrote: Hi, did anyone try do load-testing on asterisk, for sip channel calls? I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle? Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with typical codecs, is that right? Thanks in advance, Nitin ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Deployment for less than 10 phones
Im doing some research about how to deploy asterisk in small offices. So far I have seen the soekris implementation with astlinux and it sounds good. Please share your comments/ideas for the following configuration: Note: Pure PBX only, no routing/firewall functions needed. Small Office #1 Up to 10 analog phones (FXS) Up to 3 or 4 PSTN lines (FXO) Asterisk providing standard pbx features and voicemail.(no call center stuff) Codec is G711 Small Office #2 Up to 10 Voip Phones (sip) with g711 Up to 3 or 4 incoming SIP lines via Ethernet from the VOIP provider Asterisk providing standard pbx features and voicemail. No PSTN connectivity (or maybe just one emergency port???) The idea to use G711 is to minimize transcoding and to maintain the costs to a bare minimum. Either using a standard PC or a soekris board (Epygi Quadro is too expensive and I dont need the routing functions). Usually is accepted that using G711 on each leg, it needs 30MHZ per voice channel so a 300MHZ computer will give me the 10 calls I need while keeping the CPU transcoding to a minimum. Soekris boards/case cannot fit a TDM400 card unless that has changed recently, Any ideas if sangoma cards fit? Also, the net4801-60 soekris board has a 266MHZ cpu so i will only get about 8 calls. However I need some light here8 calls FXS to ZAP? SIP to SIP? Suggestions for small form factor cases are welcomed. Thanks for all your comments. Thanks for our comments. -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk 1.2.10 and Zaptel 1.2.7 released!
Matt, What do you mean the 1.2 svn branch? Where are the download instructions and installation procedure? I always download tar.gz (that means the official release) but i always question what do I do to keep my installation with the latest bug fixes. Thanks, On 7/15/06, Matt Riddell (NZ) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Julian Varanini wrote: What is the best way to update from 1.2.9 to 1.2.10? If it was downloaded from SVN then you can just type make update in the directory. If it was a .tar.gz download then you will need to reinstall. I would recommend using the 1.2 branch of SVN as it means you don't have to wait for the releases to get the bugfixes. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEuLm8S6d5vy0jeVcRAk9RAJ478UyMx8g7WLzkhAp+9VT9eZfXewCggHXo 9bn2Ob7u9jlDsqrKLZVrv/4= =y79J -END PGP SIGNATURE- ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] compiling zaptel 1.2.7 error: stray '\194' in program
Hi, While compiling zaptel 1.2.7 I get the error: zaptel.c:384 error: stray '\194' in program It looks like an unprintable character, this source is straight from the .tar.gz release. Since im compiling in a Centos 4.3 Xeon, the only thing I added was: CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo -Drw_lock_t=\rwlock_t\; fi) Suggestions? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: compiling zaptel 1.2.7 error: stray '\194' in program
My mistake. Adding the \ character using ascii codes in a badly configured console+keyboard, led to the insertion of a dot and that was the unprintable char. Note to console users (CENTOS?) vi and nano did not showed the char, I then used MC (midnightcommander) and it showed the dot. See ya, On 7/15/06, Erick Perez [EMAIL PROTECTED] wrote: Hi, While compiling zaptel 1.2.7 I get the error: zaptel.c:384 error: stray '\194' in program It looks like an unprintable character, this source is straight from the .tar.gz release. Since im compiling in a Centos 4.3 Xeon, the only thing I added was: CFLAGS+=$(shell if uname -r | grep -q 2.6.9-34.EL; then echo -Drw_lock_t=\rwlock_t\; fi) Suggestions? -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] How to create or test tone configuration to include them in zaptel
Hi, I would like to know what kind of tests should I make in order to document tone/configuration settings for analog cards and E1 cards specifically for my country (Panama). For example: Australia, Venezuela, etc, they have been documented and included in the zaptel config. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] OT: 3Com 3C10222 POE 24 Port Ethernet
Thanks. Now I know why was only at 100$. On 7/14/06, Jared Valentine [EMAIL PROTECTED] wrote: 3C10222 was a pre-standard 24v PoE injector. It would only power other pre-standard 24v devices such as old 3Com phones. Jared Valentine [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, July 12, 2006 6:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: 3Com 3C10222 POE 24 Port Ethernet Erick Perez wrote: There is an old, very old document that I found somewhere that this PoE switch was designed for NBX phones at that time. Does anybody in this list is using this switch with non-3com NBX PoE phones? just check the voltage specs. I think you will fry anything other than an old 3com phone. Now I believe they use the standard PoE in their new switches. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1000s of extensions in one context?
My working experience with 100s of extensions, usually associated to
personnel that will *not* change from my defaults is:
; Extensions
exten = 1000,1,Macro(call-sip-local,1000,SIP/1000,default) ; Operator
exten = _1XXX,1,Macro(call-sip-local,${EXTEN},SIP/${EXTEN},default)
Then,
[macro-call-sip-local]
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - voicemailcontext
;
exten = s,1,Set(LANGUAGE()=en)
exten = s,n,Playback(pls-wait-connect-call)
exten = s,n,Set(LANGUAGE()=es)
exten = s,n,Dial(${ARG2},20,tT) ; Ring the
interface, 20 seconds maximum
exten = s,n,NoOp(${DIALSTATUS})
exten = s,n,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION$
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED]) ; If
unavailable, send to voicemail w/ unavail announce
exten = s-NOANSWER,n,HangUp()
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED]) ; If
busy, send to voicemail w/ busy announce
exten = s-BUSY,n,HangUp()
exten = s-CHANUNAVAIL,1,PlayTones(congestion)
exten = s-CHANUNAVAIL,n,Wait(2)
exten = s-CHANUNAVAIL,n,StopPlayTones()
exten = s-CHANUNAVAIL,n,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,n,HangUp()
exten = _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
;
; end;
On 7/12/06, Roger Schreiter [EMAIL PROTECTED] wrote:
Dovid Bender schrieb:
several thousand extensions or several extensions called 1000 ?
Several thousend extensions.
exten = 497111234,1,goto(...)
exten = 497111235X,1,goto(...)
exten = 497111236XX,1,goto(...)
exten = 497111237,1,goto(...)
Several thousend extensions of maybe different length.
For overlap dialing to operate correct (and no need to
wait for timeouts) I would like to put the whole dial
plan into the file extensions.conf.
Before starting, I would like to know, whether there are
experiences with such long dialplans.
Roger.
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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Re: [asterisk-users] an operational scenario
Why can't you do it? I have an internal 192.168.100.x address (eth0) and 200.x.x.x (eth1) interface. Internal users register to the 192 and internet users register to the 200.x address internal extensions are 1XXX and external extensions are 2XXX What errors do you have? On 7/12/06, Bruce Ferrell [EMAIL PROTECTED] wrote: I'm trying to do something I've not see written up here before. I have an asterisk on a box with 2 interfaces like the drawing below. I want to have SIP extensions regsitering to both interfaces and able to communicate. Is this possible? What suggestions do you have? +-+ | | internal | | external --+ +- 192.168.1 | | real IP | | +-+ ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk stops abruptly
enable debug logging in /etc/asterisk/logger.conf then do a logger reload then if asterisk dies, search the log for relevant events and post it here. I'm also using 1.2.9.1 so im interested. On 7/11/06, Dan Brummer [EMAIL PROTECTED] wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Thank you for your help, Dan ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] several asterisk servers questions
We have a 3 stage implementation plan with a customer, we're still documenting the structure, it is of course subject to change so I welcome all your comments on this matter. customer has one main building and 5 more. All inter-connected with fiber optics links for data/voice traffic. Main building holds the datacenter. proper network gear (switches and routers) are being deployed by another company to the customer. On stage one, the customer wants to have a FAX server. They read about using asterisk as a fax server and also read about Hylafax they also read about astfax+trixbox. The setup must compete against MS windows solutions that do fax-to-email and email-to-fax, we must keep deployment of software to the client machines to a minimum. The customer is looking to deploy an E1 so faxes have 30 channels to receive and send faxes, the server must communicate with an MS Exchange 2003 server. On stage two, there will be an asterisk server to handle PSTN calls (in and out)using E1 lines /about 4 E1s. We think that due to the load (500+ SIP users in main building) voicemail should be handled by a different server. Then On stage 3, another server???, serving SIP users in the main building to connect to the other buildings that will also have a little less powerful IP-to-SIP and/or IP-to-FXS asterisk (those server may have PSTN connectivity). Some form of config backups and/or disaster recovery plans must be documented as well as taking images of the RAID systems that will be using asterisk. I'm expecting full server details on this one, because the customer will provide the equipments (servers). So your comments will be appreciated. Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] Asterisk stops abruptly
no. just logger reload On 7/11/06, Dan Brummer [EMAIL PROTECTED] wrote: Thank you for the quick response. I assume this change will require an Asterisk reload? Thanks! -Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Tuesday, July 11, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk stops abruptly On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote: Hello, I'm recently having the problem where Asterisk just stops working. The console gets disconnected and the process appears to die. I am using Asterisk version 1.2.9.1. Anyone have any ideas on where I should be looking for the cause of my problem? Also, I notice there is a /var/log/asterisk/messages log file but it doesn't contain any information that I can use to help troubleshoot the application crashing. Is there a way to put more debugging in the log file? Yes take a look at logger.conf. There is a default of 'full' which will create /var/log/asterisk/full for example, and will have more info, but you can add the individual elements to the messages one if you would rather. -- Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058 US WA +1 360 207 0479 US NY +1 516 687 5200 FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
One of my customers decided to allow me to make a test system for a fax server. So far I have searched the wiki and came up with Hylafax(standalone or with IAX) and astfax (integration with asterisk). Scenario: Customer has windows machines (500+) and we want to try a fax server in the email-to-fax fax-to-email mode with minimum intrusion in the windows machines. astfax looks promising but it uses openoffice libraries to do conversion from .doc or other formats to tiff. The thing with this is that OO sometimes lacks the reliability to do a true conversion on MS Office formats like fonts or spacing or tabs. So it will look good on MS Word for example, but crap after OO conversion. I have no intention to start a war on this, but those who use MS Office and OO will know that true font/spacing/etc conversion is far from perfect, specially when mixing different MS Office version (95,2000,XP) AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. Any suggestion for this kind of setup? Thanks, -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
We used? what are you doing different now? On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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] setting up an email to fax with asterisk
Hehe, ok. Thanks, On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: A different job PaulH On Mon, 2006-07-10 at 03:34 -0500, Erick Perez wrote: We used? what are you doing different now? On 7/10/06, Paul Hales [EMAIL PROTECTED] wrote: We used to put one of the hylafax printer drivers on each windows box - which is not much fun. PaulH On Mon, 2006-07-10 at 02:52 -0500, Erick Perez wrote: So I install a machine with linux+hylafax+asterisk+iaxmodem+TDM400 (4fxo) and then how the windows clients send email-to-fax to the above machine? On 7/10/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jul 10, 2006 at 02:35:23AM -0500, Erick Perez wrote: AS with Hylafax, it seems that I need to install an IAX modem in every machine (arrrggg) or define a printer driver. You need to install an iaxmodem on the machine where the hylafax server is installed. Which can probably be the Asterisk server. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.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 ___ --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 -- Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ___ --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
