Re: [asterisk-users] Asterisk 1.4 with Unicall
Hi Carlos, On 6/8/07, Carlos Chavez [EMAIL PROTECTED] wrote: I have a small call center running with Asterisk 1.4.4 and Unicall. Everything seems to be working but twice now we had to reset the server because all lines stopped working. You can see users dialing in and reaching the queue but the agents never get the call and the lines are not released. Hum... do you have all the logging facilities enabled? some debug/warning messages will help out with this. I would like to know more details about what do you mean with stop working. R2 signaling failures? ( I dont think so ) I saw that there is a new Zaptel driver which fixes a racing condition with a TE110P card which is what we are using. Could this be the problem? No idea, but you should try the new driver [Jun 8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 16 [Jun 8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate: unicall_indicate -1 [Jun 8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 16 on channel UniCall/1-1 [Jun 8 18:36:02] NOTICE[16202]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 16 [Jun 8 18:36:02] WARNING[16202]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 16 on channel UniCall/1-1 [Jun 8 18:36:02] NOTICE[16350]: chan_unicall.c:2401 unicall_indicate: unicall_indicate 17 [Jun 8 18:36:02] WARNING[16350]: chan_unicall.c:2449 unicall_indicate: Don't know how to set condition 17 on channel UniCall/14-1 What do they mean? I've never seen them under Asterisk 1.2 Condition 16 and 17 are used for Hold and Unhold events. This only means the unicall driver is not handling those events. Im not sure if we should be handling those ones somehow in the Unicall driver. Do you hear music when putting the Unicall channels on hold? if not, then I think we should be calling ast_moh_start() just as other drivers do. Steve have made some improvements to unicall so im working in doing the proper modifications to Asterisk Unicall driver in order to work, so I will add this onhold thing to my TODO. Regards, -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ --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 -- 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
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Alvaro... Hum..., I never have tried RxFax... let me know if you need any extra help with that. Sounds interesting On 6/8/07, Alvaro Parres [EMAIL PROTECTED] wrote: Moy: I have working an Asterisk 1.4.4 with Unicall rn MFR2. The only problem i have is the RxFAX application, that broke every time... With and error in the linking to the spandsp library. If i have time this weekend i will review to fix the app, Thanks. On 6/4/07, Tobias Wolf [EMAIL PROTECTED] wrote: Humberto Figuera schrieb: HI Tobias, look in www.soft-switch.org/unicall/unicall/index.html ;p Thank you. Not very complete but it has given me an idea what to think of unicall. ___ --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 -- 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
Re: [asterisk-users] Bad Echo between SIP calls
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. 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
[asterisk-users] Asterisk Users Conference Friday: New Asterisk Book and a visit from JerJer of Nufone
Oops, I had some problems and was offline unable to remind you about the conference yesterday. LISTEN to recent recordings: http://x2z.eu/astusers.htm (Flash player, will autostart) THIS WEEK: Stephan Winterberg and Stephen Boche tell us more about the new book, whick looks like a great effort. A surprise visit from Jeremy, one of the pioneers of our community who started Nufone when someone on IRC said I need a new phone. SIP call instructions: http://x2z.eu/ Download mp3 or listen to any conference recording here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 Suggestions for June22nd and 28th are welcome. I have spoken to a couple of people about the 22nd but have no confirmations yet. randy ___ --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 MS RTC Library Ethernet Capacity
On 08/06/07, Asterisk [EMAIL PROTECTED] wrote: Would a good 1 gBit switch be enough to handle that (Asterisk box would be connected to that switch with 1 gBit connection, and computers with Microsoft RTC Library would be connected with a 100 mBit connection)? Alex: 30 concurrent calls will be about 2.4 megabits in each direction. 10baseT would probably just about handle it. 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
Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies [EMAIL PROTECTED] wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. 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 - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --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] Bad Echo between SIP calls
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Saturday, June 09, 2007 4:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve Most likely the phones. Is it worse on speakerphone? Are they cheap like the Grandstream 101s? Try with a couple softphones and headsets, any better. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --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] Bad Echo between SIP calls
Do you have reinvites enabled? Are you running this over a linksys four port SoHo router/switch or something? Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Saturday, June 09, 2007 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies [EMAIL PROTECTED] wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. 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 _ Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU . ___ --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] Bridged PRI calls - processor involvement?
Are you running recording on your box or FTPing large recording files or PDFs or anything other than just voice traffic? Has voice traffic spiked in conjunction with your problems? Are you doing any kind of port monitoring/mirroring on your switch? Most people look at the 100mb or 1Gb figure but there is also another very important spec to look at when evaluating a switch. It is Frame Forwarding Rate measured by Mpps. Take a look at your switch's docs and let us know what your FFR is and if you are doing any mirroring or link aggregation. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: Friday, June 08, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? It probably did but we run in updates every week and nobody can state exactly when the problem started only a few weeks ago - not very helpful. I can see that when I hear the issue the iowait time is high on the processor. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? Did it accompany an update you made? If you can find out what version the problem started occurring, that would help in fixing the problem. Matthew Fredrickson Software/Hardware Engineer Digium, Inc. On Jun 8, 2007, at 2:59 AM, Steve Hanselman wrote: The setup. Asterisk is on a 3G Zeon Dell 2850 running Fedora Core 5/6 (all yum updates applied), the TE410 lives on it's own interrupt. Asterisk sits between our telco and a PRI enabled PBX. These are the relevant versions installed: Linux: 2.6.20-1.2316.fc5smp Zaptel: 1:1.4.2.1-34.fc5 Asterisk: 1:1.4.0-34.fc5.at Libpri: 1:1.4.0-16.fc5.at Wildcard details: Found TE4XXP at base address fe3ffc00, remapped to f88bec00 TE4XXP version c01a016a, burst OFF, slip debug: OFF Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x377bb400 Reg 1: 0x377bb000 Reg 2: 0x Reg 3: 0x Reg 4: 0x0001 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1f00 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x004a TTE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE410P (3rd Gen) TE4XXP: Span 1 configured for CCS/HDB3/CRC4 TE4XXP: Span 2 configured for CCS/HDB3/CRC4 The problem: At random points during calls we lose 1-3 seconds of speech (both ways both callee and caller), this can be replicated (or at least a very good approximation!) by generating a high level of interrupt/cpu activity (for instance copying data from a USB caddy as we tried the other day in an attempt to reproduce this more reliably). The calls are bridged PRI:PRI calls, no VOIP involvement. This was not a problem until approx 3-4 weeks ago, but I can't tie it down to an exact date. Steve Interrupt sharing is not a problem anymore with those cards. What version of zaptel did you try installing? Can you explain more about your problems? Also, your configuration and setup would help out as well. --- Matthew Fredrickson Digium, Inc. The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --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 The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review,
[asterisk-users] Is There any Asterisk TODO(Developer side) list Available
Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online. http://smallbusiness.yahoo.com/webhosting ___ --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] R2 Argentina
Dear Folks, I have found that Argentine variant ar libmfcr2.0.0.3 is not set correctly Regarding ANI restriction signal. Argentine regulations since 1999 have swaped SIG_12 with SIG_15 in order To restrict ANI presentation to the user. I dont know if it has been patched in later releases of mfcr2 lib but this Simple patch works for me in mfcr2.c: /* * patch de Oscar Carriles * mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_12;*/ mfcr2-group_i_end_of_ANI_restricted = R2_SIGI_15; mfcr2-group_i_end_of_ANI = R2_SIGI_12; Thanks to Steve UnderWood for his excelent work! Ing. Oscar Andrés Carriles Director de Ingeniería Eolix Technologies S.R.L. Tel 54 11 50 32 33 52 ext. 2002 www.eolix.com.ar AVISO LEGAL: Esta información es privada y confidencial y está dirigida únicamente a su destinatario. Si usted no es el destinatario original de este mensaje y por este medio pudo acceder a dicha información por favor elimine el mensaje. La distribución o copia de este mensaje está estrictamente prohibida. Esta comunicación es sólo para propósitos de información y no debe ser considerada como propuesta, aceptación ni como una declaración de voluntad oficial de Eolix Technologies. La transmisión de E-mails no garantiza que el correo electrónico sea seguro o libre de error. Por consiguiente, no manifestamos que esta información sea completa o precisa. Toda información está sujeta a alterarse sin previo aviso -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Steve Underwood Enviado el: Lunes, 24 de Julio de 2006 07:33 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Clocking Multiple T1 Cards Andrew Kohlsmith wrote: On Monday 24 July 2006 12:11, Shaw Terwilliger wrote: Thank you; this is the kind of information I was looking for. The wiki and other documents told me exactly what the configuration options did, but I didn't know what kind of timing configuration was right for multiple cards. Essentially the timing is ONLY for the hardware on the card. The Digium cards use a quad framer chip (maybe a dual for the TE210 but I don't think so) and it's a hardware limitation of the framer that all spans must share the same clock source. Sangoma's cards use individual framers and don't have this limitation. (essentially I think it was a cost/space tradeoff.) Once the data is on the PCI bus, the clock source is irrelevant. They're all close enough that it doesn't matter anymore. This statement is very very wrong. The timing matters enormously. If the timing doesn't match, there will be frame slips, and things like modems will not work. The snag is, right now neither Asterisk or the cards it uses have the ability to lock their clocks together. Those framers want exact lock-step timing though, which is why your clocking settings are so very important, and why with telephony in general it is crucial to think about your clocking before throwing hardware at a solution. 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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/396 - Release Date: 24/07/2006 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.13/842 - Release Date: 09/06/2007 10:46 a.m. ___ --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] No sound, problem is not a NAT
HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: Connected to Asterisk SVN-branch-1.2-r68526 currently running on hernandezz-laptop (pid = 6970) Verbosity is at least 10 -- Executing Macro(SIP/5000-081da408, user-callerid|) in new stack -- Executing NoOp(SIP/5000-081da408, user-callerid: device 5000) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?start) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, REALCALLERIDNUM is 5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSER=5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSERCIDNAME=hnmvbn) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing Set(SIP/5000-081da408, CALLERID(all)=hnmvbn 5000) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?continue) in new stack -- Executing Set(SIP/5000-081da408, __TTL=64) in new stack -- Executing GotoIf(SIP/5000-081da408, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/5000-081da408, Using CallerID hnmvbn 5000) in new stack -- Executing Wait(SIP/5000-081da408, 2) in new stack -- Executing Macro(SIP/5000-081da408, systemrecording|dorecord) in new stack -- Executing Goto(SIP/5000-081da408, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/5000-081da408, /tmp/5000-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') hernandezz-laptop*CLI Really at a stand still until I can get this resolved so any thoughts are much appreciated. -- Carlos Jerónimo ___ --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] Linksys 941/942 reboot and persistent MWI
We've got a bunch of Linksys 941/942s and have them all configured to upgrade the config periodically. Problem is, when the phone loads a new config it goes through what appears to be a soft reboot, although it only takes about 5 seconds. During this time, the display goes blank and the (normally) green line buttons flash off briefly. This is a minor nuisance and elicites questions and complaints from users. But, worse, is that when the phone goes through this recycle, the red MWI light comes on. About 95% of the time, it eventually goes off by itself, but occassionally it takes a power cycle to do it. We are running the latest Linksys firwmare. My question is this. Has anyone else experienced this problem and if so, what have you done about it? I can't believe we're alone, as there must be a bezillion of these phones connected to Asterisk systems. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ --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] IAX2 Trunk No Sound
3. Can you post some of the CLI errors you mentioned? iax2_trunk_queue: Maximum data space exceeded and once this start it never gets stopped so I've to kill the asterisk and restart the whole box. Instead of restart whole box if I just try to restart the asterisk my agents not able to hear any voice. It looks like you are trying to get too many calls down a single trunk. Try defining a few trunks and split your calls over them. There is a limit on the total packet size in a IAX trunk. Or, if you have enough bandwidth between the offices, don't use trunking. - 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
Re: [asterisk-users] No sound, problem is not a NAT
What about the RTP ports (rtp.conf). Aren't they blocked on your firewall/iptables? Rgds, Ricardo Martins. Carlos Jerónimo escreveu: HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. But this isn't just voicemail. It's every system recording. Such as the feature code *60 to play the current time. It shows the call connected and it shows to be playing the wav file, but nothing coming out of the speaker of the phonedidn't just try with one phone either In other words, asterisk shows it's all working well. my logs: Connected to Asterisk SVN-branch-1.2-r68526 currently running on hernandezz-laptop (pid = 6970) Verbosity is at least 10 -- Executing Macro(SIP/5000-081da408, user-callerid|) in new stack -- Executing NoOp(SIP/5000-081da408, user-callerid: device 5000) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?start) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, REALCALLERIDNUM is 5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSER=5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSERCIDNAME=hnmvbn) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing Set(SIP/5000-081da408, CALLERID(all)=hnmvbn 5000) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?continue) in new stack -- Executing Set(SIP/5000-081da408, __TTL=64) in new stack -- Executing GotoIf(SIP/5000-081da408, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/5000-081da408, Using CallerID hnmvbn 5000) in new stack -- Executing Wait(SIP/5000-081da408, 2) in new stack -- Executing Macro(SIP/5000-081da408, systemrecording|dorecord) in new stack -- Executing Goto(SIP/5000-081da408, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/5000-081da408, /tmp/5000-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') hernandezz-laptop*CLI Really at a stand still until I can get this resolved so any thoughts are much 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
Re: [asterisk-users] No sound, problem is not a NAT
Hi Carlos - HI, my problem is with internal sounds of asterisk. for example when calling voicemail, no system recordings are being played back. However, when running asterisk in a debug mode, i see the call coming through to the system and the system playing back the wav files promptly. However, no sound comes through. I have verified that the sounds are in the correct location and that asterisk:asterisk has access to all files, is music on hold works, but other than that no system recordings are audible. Connected to Asterisk SVN-branch-1.2-r68526 currently running on hernandezz-laptop (pid = 6970) Verbosity is at least 10 -- Executing Macro(SIP/5000-081da408, user-callerid|) in new stack -- Executing NoOp(SIP/5000-081da408, user-callerid: device 5000) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?start) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, REALCALLERIDNUM is 5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSER=5000) in new stack -- Executing Set(SIP/5000-081da408, AMPUSERCIDNAME=hnmvbn) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?report) in new stack -- Executing Set(SIP/5000-081da408, CALLERID(all)=hnmvbn 5000) in new stack -- Executing Set(SIP/5000-081da408, REALCALLERIDNUM=5000) in new stack -- Executing NoOp(SIP/5000-081da408, TTL: ARG1: ) in new stack -- Executing GotoIf(SIP/5000-081da408, 0?continue) in new stack -- Executing Set(SIP/5000-081da408, __TTL=64) in new stack -- Executing GotoIf(SIP/5000-081da408, 1?continue) in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp(SIP/5000-081da408, Using CallerID hnmvbn 5000) in new stack -- Executing Wait(SIP/5000-081da408, 2) in new stack -- Executing Macro(SIP/5000-081da408, systemrecording|dorecord) in new stack -- Executing Goto(SIP/5000-081da408, dorecord|1) in new stack -- Goto (macro-systemrecording,dorecord,1) -- Executing Record(SIP/5000-081da408, /tmp/5000-ivrrecording:wav) in new stack -- Playing 'beep' (language 'en') hernandezz-laptop*CLI I'm a little confused. Your CLI output doesn't show asterisk trying to play anything. What happens if you set up an extension that just does something like: exten = 100,1,Playback(vm-instructions) Do you get sound? You didn't mention - do you get normal audio on calls between your SIP phones? - 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
[asterisk-users] Re: SIP NAT ...
On Fri, 1 Jun 2007, Gordon Henderson wrote: So I thought I had SIP and NAT cracked a long time ago, but something's just happened that's sort of upset the cart )-: I have an * box behind a NAT firewall. Nothing unusual there, this is something I've done many times - sip.conf has the correct nat= localnet= externip= settings, the router has ports 5060-5069 and 1-2 forwarded to the internal IP address of the * box. (and 4569 for IAX, but we're just using SIP here) The * server has a few internal (LAN) and external SIP phones, but also has 2 SIP connections to an external PSTN provider. I don't know what this is as I don't have any control or access to it, but both go to the same IP address with different account details (username/passwords) Both these SIP - external PSTN provider connections register OK on the * box, and outgoing calls placed over either connection works perfectly. Outgoing callerId (set by the external provider) works as expected. ) I have dialling prefixes for each 'line', nothing special there, that side of it all works as expected. The problem is that only the last one in the sip.conf file actually accepts incoming calls when dialled from the PSTN side. (They have different PSTN phone numbers) If I swap their entries over in the sip.conf file, then the other one takes the calls. When dialling the first number, nothing seems to get through to the * box at all - nothing on the console in verbose mode, nothing in the log-file. The 2 SIP account setups are otherwise identical (generated by a web interface), just the usenrname password differing, and the account name. So I have now had to time to get to the bottom of this and it seems to be nothing to do with NAT. I even found some pages on the WiKi that were useful: (Although not under any heading I'd have thought of!) http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer or at least talked about others having the same problem and presents some solutions one of which I managed to adapt for my situation. I'm not sure if I'd call this a bug, feature or just a PITA )-: Incoming SIP calls (from registered hosts, when we're a peer?) start from the bottom of the sip.conf file and work upwards. However the bottom entry would appear to be the one that's used to authenticate all of this type of incoming SIP calls, the others aren't. So I created a new peer entry at the bottom of the file, insecure=invite, and a new context to send incoming calls to and it seems to work. The far-end is OpenSER in-case anyone knows of that having issues. And while it works, I don't think it's too satisfactory a solution, (and I can't work out how to put this into a GUI yet!) but there you go. Gordon ___ --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] Bad Echo between SIP calls
Yeah I have made sure its the correct port. We have 75 polycoms currently. ? the SIP-to-SIP echo is there. -- Deepak Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. Make SURE you have the handset plugged into the handset port of the phone, not the headset port of the phone. ___ --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 - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --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] Bad Echo between SIP calls
reinvite is disabled. Also its a Dell PowerEdge 850 server running asterisk
connected to a Cisco switch. other network in company have Cisco Switch.
Also we have approx 75 Polycoms all over.
canreinvite=no
--
Deepak
Steve Totaro [EMAIL PROTECTED] wrote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) }Do you have reinvites
enabled? Are you running this over a linksys four port SoHo router/switch or
something?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu
Sent: Saturday, June 09, 2007 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior
upgrading to PRI we were till date using 4 analog lines connected with TDM card
from digium no echo for pure SIP to SIP lines.
Now I have TE212P which had onboard echo cancellor.
I am trying make myself clear before I blame on any network. B'cos for
sure we have a spegati of networks no QoS. Also the intresting thing is if I
call from one extension to other dialing the main line then extension the
call is crystal clear. but when dialing a direct extension its a hell of echo.
--
Deepak
Stephen Davies [EMAIL PROTECTED] wrote:
On 09/06/07, Deepak Naidu wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo on pure SIP to SIP calls, there's no point in
fiddling around with your zapta.conf. That file is for configuring
chan_zap, which is used to talk to Zap/ channels. Your calls are SIP
to SIP so the zap channel and your PRI aren't being used at all.
SIP calls are pure digital 4 wire lines so no electrical (Hybrid)
echo will be present. The phones should not generate echo. If they
are, they are presumably nasty phones (what kind are they?) and you
should get properly made phones.
Steve
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Re: [asterisk-users] Bad Echo between SIP calls
Stephen Davies wrote: On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. 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
Re: [asterisk-users] Hot GXP-2000
One of the reasons why I stand clear of Grandstream - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk asterisk-users@lists.digium.com Sent: Friday, June 08, 2007 6:47 PM Subject: [asterisk-users] Hot GXP-2000 ___ --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] H.323 trunk between MD110 and Asterisk
Hi. Anyhone have any experience with trunking between Ericsson MD110 and Asterisk using H.323? I've tried both ooh323 and the /channels/h323 one in version 1.4.4 and 1.4.0 of Asterisk. ooh323 does not manage to establish the call (starts to ring but then disconnection when answering the call on the Asterisk end) but using the channels/h323 driver I can get the call established from MD110 to Asterisk (still does not work in the other direction) but no sound is transferred between the two. Just dead silent on both ends. I have some logs and more details if needed and if anyone is ready to listen. Would really appreciate your input on this. tnx, Baldvin ___ --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 tell what codec is used for each end of a call MD110-H323-SIP
Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you for your time and effort to respond. Baldvin ___ --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] remove
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Saturday, June 09, 2007 5:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] How to tell what codec is used for each end of a call MD110-H323-SIP Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you for your time and effort to respond. Baldvin No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.13/841 - Release Date: 6/9/2007 8:52 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.472 / Virus Database: 269.8.13/841 - Release Date: 6/9/2007 8:52 AM ___ --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] IAX trunk with dynamic IPs
Hi Noah, First of all, thanks for your help. I just want to check if I understood. If a set the TTL for 10 seconds for host.no-ip.org and configure the parameter host as host=host.no-ip.org, Asterisk will try to find the IP address of host.no-ip.org each 10 seconds? That is it? Thanks again. Ronaldo. Noah Miller wrote: Hi Ronaldo - I have a IAX trunk between two asterisk servers, both with dynamic IP and both have a DNS name associated with it. In the iax.conf file I configure the host parameter with the DNS name of the servers. Everything works fine until one of these servers get a new IP, so the other can't find its peer (the one that has just gotten a new IP). If I manually issue a iax2 reload in the CLI, asterisk tries to find the IP of the peer (based on its DNS name) and everything starts working again. This is the section for my trunk in one of my servers: [sometrunk] type=friend username=someusername secret=somesecret auth=plaintext host=host.no-ip.org context=incoming peercontext=incoming qualify=yes trunk=yes Is there any way to tell asterisk to try to find the peer's IP address if that peer is unreachable or each 10 minutes? I don't know if your DDNS provider would support this, but if you set the TTL value of your DNS hostnames to something very low, like 10 seconds, it would force your OS to keep finding the latest IP. - 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 ___ --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] IAX trunk with dynamic IPs
On 6/9/07, Ronaldo Z. Afonso [EMAIL PROTECTED] wrote: First of all, thanks for your help. I just want to check if I understood. If a set the TTL for 10 seconds for host.no-ip.org and configure the parameter host as host=host.no-ip.org, Asterisk will try to find the IP address of host.no-ip.org each 10 seconds? That is it? I believe this sort of thing came up a few weeks ago on the list. What I remember the outcome being was that Asterisk currently will only do the DNS lookup when you initially start or do a reload from the Asterisk CLI. So, modifying the TTL on your DNS records will have no effect unless Asterisk is patched to query the DNS each time instead of relying on it's DNS cache. -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX trunk with dynamic IPs
*set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --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: CallManager ANI restamp.
Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone. This is probably because the call doesn't even run through the phone; it just sends a directive to CallManager to do something else with calls bound for that peer. How can I do that? Surely there must be a way in CallManager... Thanks! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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: CallManager ANI restamp.
Alex, While we're at it, I have this Mitsubishi Magna with a computer-controlled transmission. After a voltage sag, the TCU seems to have gone to lunch. Forcing it into limp mode by pulling the TCU fuse seems to right things (although it's then duly stuck in third gear). I know this has nothing to do with Asterisk, but each time the car fails, I need to use my mobile (cell) phone to call the RACQ (our roadside breakdown organisation here in Qld, Australia). These calls cost quite a bit, per call. How can I fix the TCU? Surely there must be a way without having to use my mobile phone to get the car towed every time... Try http://cisco.com - pretty certain they support their own proprietary, very expensive kit ;-) On Sat, 2007-06-09 at 19:28 -0400, Alex Balashov wrote: Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone. This is probably because the call doesn't even run through the phone; it just sends a directive to CallManager to do something else with calls bound for that peer. How can I do that? Surely there must be a way in CallManager... Thanks! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 Cheers, Mattt. - ROMATel - VoIP made easy - http://romatel.net - SpotSafe - WiFi Hotspot solution - http://spotsafe.net There are only 10 kinds of people. Those who understand binary, and those that don't... ___ --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] IAX trunk with dynamic IPs
Hi Matt, Every time I do that, IAX stop sending the POKE messages (necessary for trunk management). Do you know what could be happening? Thanks. Ronaldo. Matt wrote: *set enable=yes in the [general] section of /etc/asterisk/dnsmgr.conf* ___ --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] OT: CallManager ANI restamp.
On Sun, 10 Jun 2007, Mattt wrote: Try http://cisco.com - pretty certain they support their own proprietary, very expensive kit ;-) If Cisco had solutions for me, I wouldn't be humiliating myself asking this question on this forum. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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] Polycom 301 vs. 330
Is the main difference between the two the full duplex speaker on the 330? -- Warm Regards, Lee ___ --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] Polycom 301 vs. 330
Lee Jenkins wrote: Is the main difference between the two the full duplex speaker on the 330? The 330 also includes two other features over the 301: - A 103x33 pixel graphical display as opposed to the 4 line x 20 char. monochrome on the 301. - Built-in 802.3af PoE. It does not support Cisco PoE as far as I know. Other differences are that the 320/330 does not support MGCP, and includes an XHTML micro-browser. One other important difference depending on your environment is that the 320 and 330 do NOT include an AC adapter. MSRP on a 5-pack of 24v AC adapters is 99.00. The street price should be less than half of that. Jeff -- Jeff Davis Netsource Consulting ___ --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] Polycom sip.cfg / voIpProt.SIP.requestValidation.x.request.y.event
Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it does (at least conceptually) but ss the x variable still referring to a server (1 or 2)? And the y var, what is it referring to? An event? Which one? Determines which events specified with the Event header should be validated; only applicable when voIp- Prot.SIP.requestValidation.x.request is set to “SUBSCRIBE” or “NOTIFY”. If set to Null, all events will be validated. Please excuse me if it's an obvious question. -- Warm Regards, Lee ___ --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] Polycom 301 vs. 330
Jeff Davis wrote: Lee Jenkins wrote: Is the main difference between the two the full duplex speaker on the 330? The 330 also includes two other features over the 301: - A 103x33 pixel graphical display as opposed to the 4 line x 20 char. monochrome on the 301. - Built-in 802.3af PoE. It does not support Cisco PoE as far as I know. Other differences are that the 320/330 does not support MGCP, and includes an XHTML micro-browser. One other important difference depending on your environment is that the 320 and 330 do NOT include an AC adapter. MSRP on a 5-pack of 24v AC adapters is 99.00. The street price should be less than half of that. Jeff Nice. I missed those points. Thank for clarifying. -- Warm Regards, Lee ___ --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] Bad Echo between SIP calls
Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood [EMAIL PROTECTED] wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. 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 ___ --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] ast_dynamic_str_thread_build_va() is defined with 6 args but only called with 5 args??
I'm having a problem with asterisk-1.4.4 dumping core under Solaris 10 with a SIGSEGV error. gdb gives this stack trace: #0 0xfebd4d0c in strlen () from /usr/lib/libc.so.1 #1 0xfec2a386 in _ndoprnt () from /usr/lib/libc.so.1 #2 0xfec2d4bb in vsnprintf () from /usr/lib/libc.so.1 #3 0x080e86de in ast_dynamic_str_thread_build_va (buf=0x8172763, max_len=0, ts=0x81482a0, append=0, fmt=0x811dc6a %25.25s %s\n, ap=0x8046f18 X'\027\b) at utils.c:969 #4 0x080890e0 in ast_cli (fd=1, fmt=0x811dc6a %25.25s %s\n) at cli.c:69 #5 0x0808c946 in help1 (fd=1, match=0x0, locked=0) at cli.c:1746 #6 0x0808ca67 in handle_help (fd=1, argc=0, argv=0x8047080) at cli.c:1773 #7 0x0808d664 in ast_cli_command (fd=1, s=0x0) at cli.c:1979 #8 0x08073d0f in main (argc=135695685, argv=0x80471f4) at asterisk.c:1384 I've noticed that ast_dynamic_str_thread_build_va is defined in utils.c on line 969: int ast_dynamic_str_thread_build_va(struct ast_dynamic_str **buf, size_t max_len, struct ast_threadstorage *ts, int append, const char *fmt, va_list ap) and it's called in cli.c on line 69: res = ast_dynamic_str_thread_set_va(buf, 0, ast_cli_buf, fmt, ap); Most interesting is that the function is defined with 6 arguments and only appears to be called with 5(?). Is this correct? Frank ___ --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] Hot GXP-2000
Why? Because of their excellent customer support in taking care of a problem? At the price of the Grandstreams compared to others, I can deal with a couple of bad apples. I can buy two Grandstream's for the price of a phone with similar features. I can deal with a lot of bad apples at that ratio. Their sidecar is so cheap compared to others it's not even funny. Plus, I can't even get some of the functionality the GXP's give me from other phones. On 6/9/07, Dovid B [EMAIL PROTECTED] wrote: One of the reasons why I stand clear of Grandstream - Original Message - From: Carlos Chavez [EMAIL PROTECTED] To: Asterisk asterisk-users@lists.digium.com Sent: Friday, June 08, 2007 6:47 PM Subject: [asterisk-users] Hot GXP-2000 ___ --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
Re: [asterisk-users] Noise on FXS ports (Sangoma)
John Novack wrote: In this troubleshooting case, it probably is better that there is NO dialtone, which would make the hiss easier to hear. I am curious what the OP found When Asterisk is stopped, does the hiss continue? Okay -- I have tested this and yes, the hiss is still present even after Asterisk has been stopped. That would help to narrow down the location of the problem It sure sounds to me as if it is a hardware problem within the card or even the module, rather than on the PCI bus or within Asterisk Well, we know it's not Asterisk now... I will have to try this hardware in another machine, or try new hardware in this machine. -Stephen- ___ --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] Bad Echo between SIP calls
The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : -- Deepak C F [EMAIL PROTECTED] wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. 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 ___ --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 - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --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] Bad Echo between SIP calls
It doesn't matter if it's supported, they are all, however I have seen some echo problems after firmware upgrades, the only way to fix it was to either copy the differences or overwrite my old config files with the new ones that came with the firmware and then modify as needed for my setup. On 6/10/07, Deepak Naidu [EMAIL PROTECTED] wrote: The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : -- Deepak *C F [EMAIL PROTECTED]* wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. 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 ___ --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 -- New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championshipshttp://uk.rd.yahoo.com/mail/uk/taglines/default/championships/games/*http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/. Plus: play games and win prizes. ___ --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
