[asterisk-users] Possible Packet loss but need an opinion
I have an Asterisk box at a clients site, and they are having quality issues, they are using g711 for the codec and I think it might be that asterisk is just overloaded. Upstream I did a packet capture. The capture showed traffic going to the asterisk box from the ISP is clean, but traffic from the asterisk box to the ISP has packet loss. So we moved farther and farther down the chain until we got to the box itself. We then did a capture on the asterisk box directly using tcpdump. We found that it showed the same results. Packetloss leaving the box, no packet loss coming into the box. What would this suggest? At most we've had 150 concurrent streams, We are running the Business version. Anyone have any ideas? Thanks Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Graphing Jitter Packet loss and Latency for SIP Calls
On Fri, Apr 25, 2008 at 2:55 PM, Vikas [EMAIL PROTECTED] wrote: B. Network between the SIP endpoints and VOIP server: The Indian office has 5 different ISPs giving the internet connection. Each ISP has a different packet loss latnecy and Jitter and these change over time. So I want a way to be able to select the best ISP on a given day. I would recommend smokeping, it won't monitor the quality of the call, but it will give you a good idea of how the circuit performs. ___ -- 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] Linksys/Sipura SPA-942 phones in larger deployments
On 3/22/07, Bryan M. Johns [EMAIL PROTECTED] wrote: Chris, Having deployed every major brand of sip handset in numbers greater than 100, I can say that I recommend the Polycom product hands-down for these types of roll-outs. Provisioning and management are superior and the product if of generally better quality than the SPA line. I'd have to fully agree with you. I work as a voip carrier and the linksys desk phones are just horrible. They don't update like they are suppose to. Its just a total mess. Polycom hands down is the best. But aastra has been coming in strong. I have a 480iCT on my desk and a polycom 601. Both are great phones. Drew Matthews Velocity Networks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ONE WAY VOICE ONLY IN ASTERISK
I've had this exact same problem before. if you have multiple ips make sure asterisk binds to the external ip and see if this fixes it. ___ --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: Polycom compatible phone for Asterisk
I use the 301, 501, and 601 with asterisk daily. 301's are kind of cheesy and feel cheap to me but the 501 is rock solid. On 7/12/06, (AstATN) [EMAIL PROTECTED] wrote: Hi all, Can some one provide me the infor about polycom phones model that compatible and stable to work with Asterisk? I intend to purchase IP 300, and IP 501 models. Tq ___ --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
[Asterisk-Users] Re: What business IP phone to use
maybe you didn't want suggestions, but too bad :). My favorite up until recently was the polycom 501 and I found it was good quality and clear calls and priced well. but the production of te phone is slowing down so I bought a few linksysspa941. and iVll tell you I have a new favorite phone. its slick, provisioning is a breeeze and the call quality with built in qos is fantastic. I wasn't a big fan of grandstream products they seem to be cheaply made and i've had a few fail. but they do work. talking about my biased opinion I don't have onee, i'm a hobby programmer who works for a company that resells voip services and we use polycom and linksys. I just provide support for all phones so I kow how things work and don't work. I hope this helps. thanks andrew On 2/21/06, mustardman29 [EMAIL PROTECTED] wrote: I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining about it. I ended up buying a Grandstream GXP2000 and an Aastra 9133i to test so I pretty much know what those two phones are about. Lot's of people talking about Polycom phones but they still seem to have their problems and since they don't officially support Asterisk I have my concerns. I really don't want to have to keep buying phones to find out for myself as it get's expensive real fast. Is there any unbiased comparison of various phones and features anywhere. If someone wrote a book I'd buy it but it would probably be obsolete before it was published with the rate of new IP phone introductions and firmware revisons. I hear some people praising the GXP2000 phones and I gotta wonder what they are smokin (regardless of firmware revison) so I just don't know who to believe anymore. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
http://connect.voicepulse.net They support astrisk, with iax2 :)On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.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
Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's
sorry its really http://connect.voicepulse.com/On 2/14/06, andrew matthews [EMAIL PROTECTED] wrote: http://connect.voicepulse.net They support astrisk, with iax2 :)On 2/14/06, Jim Robinson [EMAIL PROTECTED] wrote: Hi Folks,Can anyone give me some good recommendations for VoIP providrs thatsupport Asterisk PBX's?We're based in Georgia and I having a hard timefinding anyoneRegards,JimPS - If you could CC me in on the reply I would greatly appreciate it! jim(-A T-)linux-sp.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
Re: [Asterisk-Users] Visual ring notification
You can check out a bunch of products for deaf people http://www.hearmore.com/store/prodView.asp?idproduct=2639idstore=1idCategory=105 THat is one in particular that looks like what you might want. Sorry for posting twice :)On 7/1/05, andrew matthews [EMAIL PROTECTED] wrote: You can get this... http://www.radioshack.com/product.asp?catalog_name=CTLGcategory_name=product_id=43-178 and if you want to get crazy you can disassemble it and change the strobe to be one of these http://homesecuritystore.com/ezStore123/DTProductZoom.asp?productID=571 they both use 12 volts and you can mount the strob anywhere.On 7/1/05, Rich Adamson [EMAIL PROTECTED] wrote: - cheap ata with telco-style industrial horn Just CYA regarding OSHA regulations on permissible noise levels.Headlines Read: three fingers missing after the Baker's handsslipped into the tomato slicer when the phone rang... ;)___ Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Visual ring notification
You can get this... http://www.radioshack.com/product.asp?catalog_name=CTLGcategory_name=product_id=43-178 and if you want to get crazy you can disassemble it and change the strobe to be one of these http://homesecuritystore.com/ezStore123/DTProductZoom.asp?productID=571 they both use 12 volts and you can mount the strob anywhere.On 7/1/05, Rich Adamson [EMAIL PROTECTED] wrote: - cheap ata with telco-style industrial horn Just CYA regarding OSHA regulations on permissible noise levels.Headlines Read: three fingers missing after the Baker's handsslipped into the tomato slicer when the phone rang... ;)___ Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] passing through MWI info from SBC
Is there alternative access to voicemail? Like web access? If there was then you can write a program to check the site every 5 or 10 minutes maybe every 30, and parse out the message listing and then send MWI to the phones if there is a VM waiting. Or You could get one of those cheep 20 dollar devices and hook it up to the serial port on the asterisk box and then you can use sty to monitor the port for changes. Then send the message that way. I'd have to research it a little more but it very possible. Just a thought :)On 7/1/05, Chris Gamble [EMAIL PROTECTED] wrote: Does this just sound worse than it is? With SBC you are out of luck, since Asterisk doesn't detect dialtone( it dials blind, sometimes too quickly for the CO to catch the first digit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS line removed due to problems.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]]On Behalf Of John NovackSent: Friday, July 01, 2005 10:42 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] passing through MWI info from SBCMike Myers wrote:Hi..I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501phones, and a couple sipura 841's for less used areas.We have 3 phone lines here.One is SBC, one Vonage,and one Voipjet...One hangup is that I can't figure out how to pass through a voicemail waiting indicationfrom SBC.This is important because my wife and herfamily all exchange voicemails with each other on theSBC voicemail system.They can leave messages for each other without having the phones ring, etc...Wehave a 2 yr old at home, and her sister has some smallkids too, so that's how they manage to send voicemailswhen they are unsure if the kids are sleeping, etc... Anyway, preserving this capability of using the SBC VMand being notified when a message is waiting iscritical for good WAF.The vonage line and voipjet line can be intergratedinto the Asterisk VM.My Nortel venture phones light the MWI if any line has VM on it, and the displaytells you which lines have VM waiting.I would loveto be able to duplicate this function on the Polycom'sand hopefully the Sipura's as well. I've looked for answers on this, but haven't foundone, hence the post.My apologies if I have missedsomething.Thanks much,MikeYou haven't missed much. With SBC you are out of luck, since Asterisk doesn't detect dialtone(it dials blind, sometimes too quickly for the CO to catch the firstdigit, resulting in wrong numbers )) or stutter dialtone either, and reportedly has had any indication of the DC status of a POTS lineremoved due to problems.Only choice would to port the number to a VOIP provider and provide theVM in Asterisk.Similar problem with Vonage VM. John Novack___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voice mail problem
Set DTMF to info that fixed my problem. On 6/30/05, Erdem HAKİ [EMAIL PROTECTED] wrote: Hi, Perhaps I'm wrong but if you use g729 with no translation (pass-thru) you can't hear voice mail. Set your codec to gsm or g711 and try again. Erdem HAKI – [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Betül Gözlükoğlu Sent: Thursday, June 30, 2005 1:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice mail problem Hi; Have a BUDGETONE-100 and using it with asterisk…Problem occurs when I dial message center…Message center does not accept tones (password for example) that I dial, Behaves as I do not dial any number and asks for the password again…Changed the DTMF Mode from "in-audio" to "RTP(RFC2833)" it works but this time, dialing internal numbers over telephony system is denied… Does anybody has any idea about correct configuration on Asterisk or Budgetone? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LiveVoip is Bankrupt
I can host here in the US, lots of bandwidth. I have all my own servers. I'd love to help.On 6/26/05, Matt Riddell [EMAIL PROTECTED] wrote:Andres wrote: So it looks like Livevoip went Bankrupt Sh1t.Looks like the Daily Asterisk News will need a new host.So, unless anyone can donate space for a custom php and mysql basedsite, it will be hosted in either New Zealand or Italy.Offers? --Cheers,Matt Riddell___http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]
http://connect.voicepulse.net pay as you go, no signup fee.On 6/27/05, John Goerzen [EMAIL PROTECTED] wrote: On 2005-06-26, Adam Megacz [EMAIL PROTECTED] wrote: Rich Adamson [EMAIL PROTECTED] writes: I've had pretty good luck with www.teliax.com I like them too, except for support.I have THREE tickets open with them that are ten days old and haven't received even a cursory we're looking into it response.It's absurd.I'm looking for someone that sells minutes in bulk like LiveVoip usedto.No monthly fee, just pay-as-you-go.It looks like Teliax charges aminimum of $10/mo, even if I use no minutes that month. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: teliax [Was: LiveVoip is Bankrupt]
http://connect.voicepulse.com sorry about thatOn 6/27/05, andrew matthews [EMAIL PROTECTED] wrote: http://connect.voicepulse.net pay as you go, no signup fee.On 6/27/05, John Goerzen [EMAIL PROTECTED] wrote: On 2005-06-26, Adam Megacz [EMAIL PROTECTED] wrote: Rich Adamson [EMAIL PROTECTED] writes: I've had pretty good luck with www.teliax.com I like them too, except for support.I have THREE tickets open with them that are ten days old and haven't received even a cursory we're looking into it response.It's absurd.I'm looking for someone that sells minutes in bulk like LiveVoip usedto.No monthly fee, just pay-as-you-go.It looks like Teliax charges aminimum of $10/mo, even if I use no minutes that month. ___Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] realtime - unable to find key
ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` int(1) NOT NULL default '1', `permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` char(3) default NULL, `restrictcid` char(1) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM ROW_FORMAT=DYNAMIC; and my extconfig.conf looks like this... [settings] sipusers = mysql,voip,sip sippeers = mysql,voip,sip voicemail = mysql,voip,voicemail extensions = mysql,voip,extensions and when my phone tries to register i get Mar 24 21:26:32 DEBUG[28397]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '5625551212' Mar 24 21:26:32 DEBUG[28397]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Mar 24 21:26:32 DEBUG[28397]: db.c:177 ast_db_get: Unable to find key '5625551212' in family 'SIP/Registry' Mar 24 21:26:32 DEBUG[28397]: chan_sip.c:1440 create_addr: Setting NAT on RTP to 524288 Mar 24 21:26:51 DEBUG[28397]: chan_sip.c:920 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' any idea's what could cause this? Thank you in advanced. Andrew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Realtime.
So this is what i've done so far... my extconfig.conf looks like this [settings] ;example = odbc,asterisk,alttable ;iaxusers = odbc,asterisk ;iaxpeers = odbc,asterisk sipusers = mysql,voip,sip sippeers = mysql,voip,sip voicemail = mysql,voip,voicemail extensions = mysql,voip,extensions and my extensions.conf has the following lines in it. [outgoing] switch = Realtime/[EMAIL PROTECTED] [asterisk1] switch = Realtime/[EMAIL PROTECTED] include = outgoing --- so now i'm trying to get calls to work correctly, and when a incoming call comes in it gives this error Mar 23 20:09:18 DEBUG[27067]: res_config_mysql.c:117 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip WHERE name = '5625551212' Mar 23 20:09:18 DEBUG[27067]: res_config_mysql.c:605 mysql_reconnect: MySQL RealTime: Everything is fine. Mar 23 20:09:18 DEBUG[27067]: db.c:177 ast_db_get: Unable to find key '5625551212' in family 'SIP/Registry' i'm stumped i'm not sure why the phone won't register. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
