Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target
On Sat, Dec 18, 2010 at 6:51 AM, Doug Lytle supp...@drdos.info wrote: John Reynolds wrote: Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John, Without seeing any of your dial plan or any of the output from your console during the failed transfer, nobody is going to be able to help. Why don't you start by posting the relevant part of your code that does the dialing and shows up the console output during a test transfer? Doug Doug and Darrick, Thanks, I work on this a bit more and get back with more info. John R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target
The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is for my wife, then I would like to transfer the call to the fxs/dahdi analog cordless phone (x 201). At one time this worked, but about a year or so ago it stopped. What is happening now is that the call comes in (x 401), is transferred via the cisco transfer soft button to (x 201), ... during this time the caller was put on hold or rather was automatically connected to the MOH process... , When (x 201) answers the phone, they are connected to the MOH process and cannot hear or talk to the original caller. In testing, if I leave the (x 201) call open, the original outside call is kept open as well (the original caller hears nothing). A look at the active sessions confirms this. When either (x 201) or original caller hang up, the call/connection is terminated. I can transfer calls from one Cisco to the other without issue. I have looked around at my configs, but don't see anything that would cause this... but truthfully I don't even know where to begin with something like this. I checked the logs to see if there was something helpful there but did not see anything. My only though is that it is something with the way the Cisco internal transfer process happens... but again, I don't know where to begin to test that theory. Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer (sip - dahdi) results in moh for dahdi
I have had this problem for a while, so I can't be sure when it started or what was changed. The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is for my wife, which it usually is, and she is walking around the house, then I would like to transfer the call to the fxs/dahdi analog cordless phone (x 201). At one time this worked, but about a year or so ago it stopped. What is happening now is that the call comes in (x 401), is transferred via the cisco transfer soft button to (x 201), ... during this time the caller was put on hold or rather was automatically connected to the MOH process... , When (x 201) answers the phone, they are connected to the MOH process and cannot hear or talk to the original caller. In testing, if I leave the (x 201) call open, the original outside call is kept open as well. A look at the active sessions confirms this. When either (x 201) or original caller hang up, the call/connection is terminated. I can transfer calls from one Cisco to the other without issue; and if my laptop, with the softphone installed, had not just taken a turn for the worst, I would test Cisco to Bria/Counterpath and let you know how that would work. I have looked around at my configs, but don't see anything that would cause this... but truthfully I don't even know where to begin with something like this. I checked the logs to see if there was something helpful there but did not see anything. My only though is that it is something with the way the Cisco internal transfer process happens... but again, I don't know where to begin to test that theory. Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: cisco 7961 , asterisk and busy lamp : solved
Max, How did you fix it? It seems like knowledge that could be shared, even if it was basic oversight. John On 11/25/06, Max Bergmann [EMAIL PROTECTED] wrote: Max Bergmann schrieb: How can i programming a Cisco 7961 to be used as busy lamp field? my configs : sccp.conf : [devices] type= 7961 tzoffset= 0 autologin = 601 speeddial = *31, Hanna -- other SIP telefon extensions.conf : exten = *31,hint,SIP/hanna exten = *34,hint,SCCP/601 on SIP Telefon ( SNOM 360 ) everything functions good and i have busy lamp when cisco telefon Offhook, but differently does not function any idea ? Any input is greatly appreciated. ___ --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 solved my problem, thank you for your help ___ --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] Now that Nufone is dead...
I finally RESPORG'd my 800# to a different company.I didn't want to do it, but I did not want to lose my number, and was not getting any respose from [EMAIL PROTECTED]. JR On 7/5/06, Carlos Chavez [EMAIL PROTECTED] wrote: On Wed, 05 Jul 2006 13:00:53 -0400, John Kington wrote At 09:29 AM 7/5/2006 +0300, you wrote: I have tollfree numbers with Nufone working OK. But what I like most is the regular numbers with charge/month but no charge/min on incoming calls... Did your tollfree number(s) with Nufone get cut-off in April? Did you keep the same number or did you signup for another number? I requested Nufone transfer my tollfree number in May and it is still not working (code is 77-4). I am wondering if this has happened to everyone or if my number fell through the cracks.No, I have 2 toll free numbers with them that have been offline sinceApril.Every time I contact support they just say to check that my contact info is correct and that someone should be calling me soon to address theissue (which they never have).--Carlos ChavezDirector de TecnologíaTelecomunicaciones Abiertas de México S.A. de C.V .Tel: +52-55-91169161 Ext 2001___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] How is Teliax ?
I agree with Dan,I have had the same expirence, and have the same set up - Junction Networksfor Inbound, Teliaxfor outbound. And as to switching to SIP instead of IAX2... I would love to try that out, but the cable modem provider I use serves me up a NAT'd IP address and IAX traverses that fine, SIP does not. HTH, JR On 3/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote: I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on Teliax before i purchase. suggest me if there are better sevice providers.I have had issues with termination on teliax. Callers tell me I sound choppy to them. Teliax origination has no problems at all strangelyenough.For termination I use junction networks. No problems with choppiness eventhough junction networks is 3x further away from me on the internet than teliax. And I have tried all the different teliax gateways. I suspect thedifference in termination may be that junction networks has better jitterbuffers than teliax.-Dan___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] NEWS: SIP Firmware Available for Cisco 7970
Anyone got this working yet?Nope :( Any update to this status? JR ___ --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] Teliax - Codec Preference effective?
Brent,I had this same problem with Teliax (atleast it sounds the same). I had wanted to use g729 over IAX, so I set that on the Teliax website, but it would not connect. After weeks of asking for resolution, I just gave up and used g726 which was working. Then, about a month later, I moved my * box and while taking the time to set it up again I thought I'd try g729 with Teliax again... This time it worked?!? I was glad it worked, but couldn't figure out why. I explained the situation to Teliax, and never got a response. I still use Teliax, and have no problems, but I don't give them high marks for how this situation was handled.JRPS - have you tried:disallow = allallow = g729 On 1/31/06, Brent Torrenga [EMAIL PROTECTED] wrote: Has anyone had problems getting their preffered codecs on the Teliax webinterface taking effect?I have two accounts, two separate yet similarly configured * servers. On oneaccount the settings took right away - on another server I am getting no result. In fact, no matter what I change the settings to, only the oldcodecs are usable (otherwise * says it can't negotiate a codec). Teliaxsupport says it is an * bug, but I think contrary... One * box works, one doesn't... Same config... Anyone else?Sincerely,Brent A. Torrenga[EMAIL PROTECTED]Torrenga Engineering, Inc.907 Ridge Road Munster, Indiana 46321-1771219.836.8918x325 Voice219.836.1138 Facsimilewww.torrenga.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Looking for the .xml file format for idleURL for Cisco 79xx
Anyone care to post the format of this file? I've been looking all over, couldn't find it on the Cisco website. I'm open to correction.Thanks for you assistance,John ___ --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] Disregard: Looking for the .xml file format for idleURL for Cisco 79xx
Got the answer on the chan_sccp list. ThanksJohn On 1/25/06, John Reynolds [EMAIL PROTECTED] wrote:Anyone care to post the format of this file? I've been looking all over, couldn't find it on the Cisco website. I'm open to correction. Thanks for you assistance,John ___ --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] wav to g729
Kris, has this module been added to Astlinux? will it be? John R. On 12/22/05, Kristian Kielhofner [EMAIL PROTECTED] wrote: Innocent Evil wrote: how? where can I get more documentation. Thanks, -- You don't have any choice, you already made it before you came here. There are instructions at the site.You will probably need to convertthe wav files to slinear before you use it:sox $INFILE -t raw -r 8000 -s -w -c 1 $OUTFILE resample -qlThat will output in the proper format for Asterisk (and res_conv). --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 ___ --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] Toll Free Providers
I have nufone.net for 800 and all seems fine... although my useage is very low. 2 cents a min. JR On 12/17/05, Tom Vile [EMAIL PROTECTED] wrote: Looking for a good toll free DID provider.Any suggestions?All ready tried Sellvoip and Gafachi and the experience was not desirable.Thanks,Tom Vile___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Small / embedded system recommendations
I use a net4801 with astlinux. works great, but I run it at home with 3 phones. I did some research before buying the net4801. Was looking at low power (watts) fanless embedded systems. One dealer I talked with suggested I look at this company: http://www.diamondsystems.com/ I have thought to buy the Morpheus board, to see what it would do with the additional power. http://www.diamondsystems.com/products/morpheus When I talked to a rep at diamond, it seems that the price was something like $350USD... I could easily be wrong. It seems like a good option, but I have no hands on expirence. (I am NOT an employee of Diamond Systems) Kristian, interested in the Morpheus for development of Astlinux? JROn 12/11/05, Alistair Cunningham [EMAIL PROTECTED] wrote: I know it's been asked before, but this area moves rapidly.Would anyone have recommendations for a small or embedded systemsuitable for running Asterisk on? Ideally, we'd like two boxes:- One using compact flash, and is fanless, with rapid booting. - One with a hard disk for voicemail, call recording, etc.Preferably they would be capable of bridging 60 calls Zap-Zap orZap-SIP, but we're willing to consider less powerful systems. Theability to take a single Digium card is desirable. Being able to run MySQL, Apache, and SER as well are essential, as it'sfor our upcoming ITSP in an office product, which uses these heavily.Speaking of which, this is moving forward. It will have all the end customer features of ITSP in a box(http://integrics.com/products/itsp/), but not billing, resellers,affiliates, calling cards, etc. We'll be looking for resellers, probably second quarter next year. If you're interested, feel free to email us,but we don't have much information yet, so will respond with a cannedemail for now. More information to follow.--Alistair Cunningham, Integrics Ltd,+44 (0)7870 699 479http://integrics.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Teliax experiences
I use Teliax. I think the sound quality is really very good. I get about an 80ms ping with them, but a 20ms ping to Junction Networks. Some how calls to/form Teliax sound better. With Junction Networks I get great customer service, with Teliax I get Okay to good customer service (depending on who is responding). I also use nufone.net and get good call quality, but no real human interaction. Fortunately, it just works as advertised. I did have a 800# provisioning problem, that a human corrected for me, it just took a little time. JR On 12/9/05, Rich Adamson [EMAIL PROTECTED] wrote: Howdy - This is my first post on the list, and from what I've seen of * I'm very impressed. I had a question regarding everybodys experience with Teliax or Broadvoice. I setup a Teliax trunk this morning, and had calls going out it in about 5 minutes(Had to get more coffee). Has anybody had any problems with them, outages, issues with dids etc?? They have been very good for us over the last six months. You might consider searching the list archives for things like this since there have been a lot of similar postings (not just teliax) over the last year or so. Very few outages, good to excellent support, pretty solid calls. They are apparently a little behind in their asterisk code as some things like iax trunking with jitterbuffer isn't supported. ___ --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] Asterisk, Small Business, and Teliax
Andrew, Personally, I think i would look at Teliax's Pay as You go plan. Unlimited simultaneous calls, pay for what you use. JR On 12/9/05, Andrew Berman [EMAIL PROTECTED] wrote: I'm a beginner here and am interested in Teliax. I own a small business and was wondering if you guys could help me out here. I'm basically looking for 6-8 telephone lines, but I notice that Teliax supports 4 simultaneous calls on their Corporate plan. So could I get two Corporate plans and set Asterisk to use both of them and then have, in essence, 8 people talking at the same time? If someone tries to call, would the phone ring busy or would it still go through? I plan on having a T1. Thanks for any help, Andrew ___ --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] A company that sells Toll Free Number in USA
If only looking for 800 numbers; http://www.nufone.net I use them for Toll Free, and they have been good for me. JR On 12/7/05, Alvaro Parres [EMAIL PROTECTED] wrote: Hi any one can recommend me a company in the USA that can sell me a Toll Free Number and send me the call via IP. Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Has anyone bought anything from Asteriskmall? your expirence?
I placed an order with asteriskmall.com ... but am wondering if I have made a grave mistake. Wondering if anyone has had any expirence with these people, of if it is just some scam site. You input is appreciated. John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cisco 7970
Jeremiah, You say you have your 7970 working great with * ... The 7970 only supports SCCP, so are you using the chan_skinny modules that come with *, or are you using the chan_sccp modules? Thanks for any response. JR On 11/8/05, Jeremiah Millay [EMAIL PROTECTED] wrote: I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, [EMAIL PROTECTED] wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax users, g729 question
On 10/8/05, Rich Adamson [EMAIL PROTECTED] wrote: I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. Rich, Thanks for taking the time to check and respond, and for your advise. I did as you said, and here is the output: pbx*CLI iax2 debug IAX2 Debugging Enabled Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00016ms SCall: 2 DCall: 0 [208.139.204.228:4569] VERSION : 2 CALLED NUMBER : 1917 (I XXX out the number) CODEC_PREFS : (g729) CALLING NUMBER : 4401 CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: John Reynolds LANGUAGE: en USERNAME: reynj FORMAT : 256 CAPABILITY : 63744 ADSICPE : 2 DATE TIME : 189432385 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 00073 DCall: 2 [208.139.204.228:4569] AUTHMETHODS : 2 CHALLENGE : 303570605 USERNAME: reynj Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00080ms SCall: 2 DCall: 00073 [208.139.204.228:4569] MD5 RESULT : 6006ef53daa295d58f5292837a4edb60 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REJECT Timestamp: 00060ms SCall: 00073 DCall: 2 [208.139.204.228:4569] CAUSE : Unable to negotiate codec CAUSE CODE : 58 Oct 10 16:18:03 WARNING[364]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec = I am I wrong in thinking that I my call is trying to connect via g729, but that Teliax is rejecting it? Thank you, JR ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Teliax users, g729 question
Chris, Thanks also for your response... That is one of the first things I checked... I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. Thanks again, JR On 10/8/05, Chris Coulthurst [EMAIL PROTECTED] wrote: Make sure you have g729 turned on from the Teliax customer panel on their website. Chris Coulthurst [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; John Reynolds [EMAIL PROTECTED] Sent: Saturday, October 08, 2005 8:59 AM Subject: Re: [Asterisk-Users] Teliax users, g729 question I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. I just tried g729 with teliax this morning. It worked fine in both directions using three different did's from them. I did have one test call where audio was one way though. During those test calls I watched the CLI and the calls definitely were g729 without a doubt. I used the exact same teliax server you show above. I'm running cvs-head from yesterday. If you do a iax2 debug, you should be able to spot which system is not compat with g729. That should lead you into diagnosing the problem a little deeper. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Teliax users, g729 question
I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. show translations verifies that the registration took place. When I place a call, having allow=g729 as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a call with g726 as the only allow line, then the call completes as desired. I realize I could just use that (g726), but it seems odd that I cannot connect using g729 when both of us support it. I have checked, double checked, triple checked my account with teliax, and made sure that the g729 box is checked for both sip and iax. I have also contacted support, they have responded, but not with anything that would be considered helpful. My question then to you all is this: Are you connecting to Teliax via g729? if so, how... what are you doing that I might be missing? Your guidance will be most appreciated. John R. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wifi phones - desk
On 10/7/05, Will Glass-Husain [EMAIL PROTECTED] wrote: Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL Will, I don't know of a specific wifi deskphone... but I have run my Cisco 7960G using a wireless bridge without noticable problem. Have you considered this approach? Have others tried this? what were your expirences? John R. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
