[asterisk-users] solution mediant 2000 with asterik configuration
I have done intergration of mediant 2000 and asterisk mysetup is [soft_ph][asterisk]---[mediant2k]E1---[mediant2k][asterisk]---[soft_ph] This is my setup i have done all configuration and it is working fine finally i have done all configuration on all devices 1) i have create accout in asterisk users for mediant 2000 registration 2) then create one extention on asterisk for all call forwarding on mediant 2k 3) then my E1 trunk send call on other mediant 2k and finally call land on other end asterisk server configuration part is tricky but if any one need help regrading this setup just post me E-mail i will help them My E-mail :- [EMAIL PROTECTED] Mobile:- +91-9818875535 Regards Satish Patel Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 21, 2007 at 01:40:39AM -0700, satish patel wrote: Dear all anyone have idea about connect asterisk with mediant 2000 audiocode configuration ... anybody have configuration about it I have no idea. But posting the same message under three different threads will not help. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.___ --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] Play dial tone withou answer
Yes Dave, We want to use to principle for the following reason. If the outbound call is not picked up, the inbound caller won't be charged for the call, because there was no answer. Arjan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: dinsdag 19 juni 2007 17:03 To: Lee Jenkins Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play dial tone withou answer Yes Lee, he could, however he doesn't want to answer the call until the call has been completed on the outbound leg. Dave On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote: David Boyd wrote: Two points, first (I believe from many previous threads, and viewing source code ) you must answer a call to place audio on the channel. second, depending on the type of access being used by the originator of the call, the carrier will not pass audio on the channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. Can't he just setup an extension to Answer() the call, play message or Ringing() and then transfer the call? ___ --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] qozap and zt_alarm_notify_no_master_change
I got a warning during zaptel compilation: qozap.ko needs unknown symbol zt_alarm_notify_no_master_change Is this critical/what am I missing? Thanks Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTCP NTP clock skew detected
somebody knows, what this mean, or how to avoid this messages? I have clock synchronized on asterisk server using ntpd. Internal RTCP NTP clock skew detected: lsr=4103127456, now=4103296271, dlsr=168820 (2:575ms), diff=5 Internal RTCP NTP clock skew detected: lsr=4103522652, now=4103656826, dlsr=134217 (2:047ms), diff=43 Internal RTCP NTP clock skew detected: lsr=4103782839, now=4104025582, dlsr=242745 (3:703ms), diff=2 Internal RTCP NTP clock skew detected: lsr=4104178035, now=4104279406, dlsr=101449 (1:547ms), diff=78 Internal RTCP NTP clock skew detected: lsr=4104833418, now=4104912422, dlsr=79167 (1:207ms), diff=163 Internal RTCP NTP clock skew detected: lsr=4105488801, now=4105568702, dlsr=79953 (1:219ms), diff=52 Internal RTCP NTP clock skew detected: lsr=4106404371, now=4106685114, dlsr=280756 (4:283ms), diff=13 Internal RTCP NTP clock skew detected: lsr=4107715137, now=410785, dlsr=140509 (2:143ms), diff=91 ___ --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] chan_features.so / asterisk 1.4.5
Hi, after updating from asterisk 1.4.4 to 1.4.5 I get a warning for chan_features.so: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by asterisk 1.4.5? Regards, Jack ___ --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] international numbers...
Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +61242110 to something like 02422110 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XX but that still seems wrong to me. it could match other numbers. anyone had to do this in the past ? 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
[asterisk-users] asterisk 0 dial outgoing call
Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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] qozap and zt_alarm_notify_no_master_change
On Fri, Jun 22, 2007 at 12:13:42AM -0700, Vieri wrote: I got a warning during zaptel compilation: qozap.ko needs unknown symbol zt_alarm_notify_no_master_change Is this critical/what am I missing? patches/zaptel.patch You need a version of zaptel patched with the Bristuff patch. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged PRI calls - processor involvement?
Tracked this down (or more to the point found the issue causing it), it was high levels of bursty disk activity. The iowait went through the roof (30-40%). The disks are scsi serviced by an MPT-Fusion controller in a Dell Poweredge 2850. We're using LVM to bind the disks into a JBOD set. Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: 11 June 2007 10:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? I checked for BIOS upgrades the other week and there were none. I'm starting to suspect kernel changes as being the reason for this so I guess I'm going to have to remove some of the patchy disk activity to smooth the load and then start researching!!! Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: 11 June 2007 09:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bridged PRI calls - processor involvement? On Mon, 11 Jun 2007, Steve Hanselman wrote: This is the io wait figure from vmstat. If I run a vmstat 2 whilst I'm on a call I can see that the wa figure gets very high when the missing audio problem occurs. I once looked after a Dell 2850 that exhibited some odd behaviour that I never got to the bottom of. It would seem to lock-up or just crawl for 2-3 seconds every now then. Nothing logged, noting on the console. It had 6 SCSI drives fitted. I rebuilt the server twice, rebuilt the s/w RAID arrays twice, even put all 6 drives in another box (which appeared towork OK), but never got to the bottom of it. Each disk would benchmark really fast individually, Ethernet performance was good, but overall, when everything was used together, it just didn't feel right. (compared to other Dells and other servers, biger smaller that I've built and used over the years). I'd see processes hung in a D state (waiting for IO to complete) for what seemed like an overly long time, (waiting on disk), but ... I suspected a BIOS pproblem, but never had a chance to get to the bottom of it. (It was a live server doing *everything* for a small company - DNS, NIS, NFS, Intranet/WiKi, Samba, etc, etc, etc,... so taking it offline for tests was problematic) So I wonder if looking at the BIOS and seeing if there are any Dell upgrades avalable for it might help? Gordon Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: 08 June 2007 19:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bridged PRI calls - processor involvement? iowait time? I'm not familiar with that. Where are you seeing that? Also, is it a reproducible problem? --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 8, 2007, at 11:23 AM, Steve Hanselman wrote: 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
[asterisk-users] Query
Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. Can anybody help me out. Thanx and Regards sanchal singh ___ --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] Config for TEI parameter
Hi, I use a isdn card with chipset HFC and now I have needed of to config the TEI parameter to 0 (alway 0 therefore must be TEI static). But what is the parameter that I must modify in config file ? Thanks. Salvatore. ___ --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 0 dial outgoing call
On 6/22/07, satish patel [EMAIL PROTECTED] wrote: Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call create dialplan for the same ram -- Get the Yahoo! toolbar and be alerted to new email http://us.rd.yahoo.com/evt=48225/*http://new.toolbar.yahoo.com/toolbar/features/mail/index.phpwherever you're surfing. ___ --Bandwidth and Colocation provided by Easynews.com http://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] problem with one way audio
I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way audio. A caller from the pstn world hits the tdm400 card, This rings two phones in a ring group. My client answers the phone, the calling party is told the customer here her but she can not here them. The customer hangs up and calls back and the call goes through.. I rolled back to 1.2.14 and the problem is much better but is still there, Are there any ideas Don Briggs 573-614-5667 ext 4037 ___ --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] Query
Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh ___ --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] Config for TEI parameter
On Fri, Jun 22, 2007 at 12:35:09PM +0200, Salvatore wrote: Hi, I use a isdn card with chipset HFC and now I have needed of to config the TEI parameter to 0 (alway 0 therefore must be TEI static). But what is the parameter that I must modify in config file ? Thanks. Use ptp rather than ptmp. For instance: with zapbri (bristuff) you should set signalling = bri_cpe instead of: signalling = bri_cpe_ptmp -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] POTS - Incoming Voice or Fax - How to tell?
Hi Folks - This may sound weird - but here goes: I live in Japan and on my home POTS line I have a Fax/Phone machine. If I receive a fax, the thing automatically switches to 'fax mode' and prints the fax. If the call is a 'voice call', it sits there rings until answered. The above is very reliable and works okay. Of course signalling differs in each country (and even by Telco supplier) but my question is: Basically, how does the machine know if the incoming call is a fax or voice call? If there's a way to tell.. Is there a way (for example) to plug the POTS line into a FXS port then plug the fax machine into the FXO port... AND... If the incoming call is a fax, let Asterisk route it to the FXO port to print the fax. If the incoming call is voice, have Asterisk send the call to one of the SIP hardphones. Of course, Asterisk would have to figure out what type of incoming call this is. Just thinking. - Is this do-able? Thanks in advance Gary Guthary ___ --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] Friday June [EMAIL PROTECTED]:30PM EDT Asterisk Users Conference
Hi, Quick reminder that the conference is happening today at 12:30 PM EDT. I'd like to talk more about updating to 1.4. I now have a test box running asterisk 1.4.5, CentOS 5 and Lumenvox speech rec software. Seems to be fine except for some double NAT issues that could be router specific. Byran Johns from Shelton-Johns is our guest to share some of his extensive experience. More about him at 12:30PM EDT. Look for the info to join at http://x2z.eu You can listen to the stream anonymously by going to this page and clicking the Listen Now link: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 Joining Talkshoe is no big deal and makes it easy for me to see who's there and call on them for questions or comments. No real identity info is required, so please join and use your PIN. For the more brave, a Java app for windoze is available to allow chat and see who's there. Otherwise use irc.freenode.net #asterisk-users-conference to chat and send in questions ___ --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 GUI
So I'll ask the question. What's wrong with top posting. I use a blackberry to read most of my email, and bottom posting means excessive scrolling, often waiting to download additional content resulting in higher usage fees and rsi on my thumb for scrolling 90% of messages including all general email conversations are too posted yet discussion groups want bottom posting. Why? Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Thu Jun 21 12:48:45 2007 Subject: Re: [asterisk-users] Asterisk GUI On Jun 20, 2007, at 5:04 PM, Troy Ayers wrote: I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob OK, Now I'm confused... I was prepared to accept Rob's argument due its beautiful, flawless logic. But Troy has a valid point: Rob did top-post, invalidating his point. But so did Troy, invalidating his point, so now I'm stuck. Whatever shall I do? I think I'll just stick with my own opinion, seeing as both Rob and Troy are obviously idiots. (duh!) ;-) Tom ___ --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] Query
On Fri, Jun 22, 2007 at 03:20:07PM +0530, [EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18 kernel. I am using kernel 2.6.18 but facing a very serious problem of modutils and iptable. What problems, exactly? While it should be possible to install the card on such a system, is there any good reason you keep using such an old and unmaintained OS? If you're used to working with the RedHat way, why not try Centos (or buy RHEL)? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote: So I'll ask the question. What's wrong with top posting. I use a blackberry to read most of my email, and bottom posting means excessive scrolling, often waiting to download additional content resulting in higher usage fees and rsi on my thumb for scrolling 90% of messages including all general email conversations are too posted yet discussion groups want bottom posting. Why? I dont know What's the answer? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.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] qozap and zt_alarm_notify_no_master_change
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Jun 22, 2007 at 12:13:42AM -0700, Vieri wrote: I got a warning during zaptel compilation: qozap.ko needs unknown symbol zt_alarm_notify_no_master_change Is this critical/what am I missing? patches/zaptel.patch You need a version of zaptel patched with the Bristuff patch. That clarifies it. Thanks Tzafrir. Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. http://new.toolbar.yahoo.com/toolbar/features/mail/index.php ___ --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] got-name
Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 0 dial outgoing call
can u give me example how do i create plan for this task or job ram [EMAIL PROTECTED] wrote: On 6/22/07, satish patel [EMAIL PROTECTED] wrote: Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call create dialplan for the same ram - Get the Yahoo! toolbar and be alerted to new email wherever you're surfing. ___ --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 - Shape Yahoo! in your own image. Join our Network Research Panel 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] Play dial tone withou answer
Hi Arjan, As I see it, the issue at hand is as follows: You are attempting to provide a tandem service, meaning as you say no charge to the originator unless the called party answers. However under this circumstance you want to also provide a non-standard call treatment to the line without an answer occurring. Standard treatment is to allow the originating Switch/device to continue to provide the ringing condition to the originators phone while the outbound attempt is being completed. Very few carriers that utilize digital services (non-analog) do not propagate audio back to the originating caller until such time as an answer has been accomplished. SO, this leads me to asking the following, how are the callers originating calls into your system, what are they using for authentication as well as indication of desired outbound calling data? Dave On Fri, 2007-06-22 at 08:22 +0200, Arjan Kroon wrote: Yes Dave, We want to use to principle for the following reason. If the outbound call is not picked up, the inbound caller won't be charged for the call, because there was no answer. Arjan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: dinsdag 19 juni 2007 17:03 To: Lee Jenkins Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play dial tone withou answer Yes Lee, he could, however he doesn't want to answer the call until the call has been completed on the outbound leg. Dave On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote: David Boyd wrote: Two points, first (I believe from many previous threads, and viewing source code ) you must answer a call to place audio on the channel. second, depending on the type of access being used by the originator of the call, the carrier will not pass audio on the channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. Can't he just setup an extension to Answer() the call, play message or Ringing() and then transfer the call? ___ --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 GUI
Let's look at your message: On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote: The problem with top-posting is that answer comes before the question. And hence you don't really know what the question was. So I'll ask the question. What's wrong with top posting. (The above answer should have been here, if I used proper quoting. I avoided it for the sole instructive porpose of demostrating the problem with top-posting) I use a blackberry to read most of my email, and bottom posting means excessive scrolling, often waiting to download additional content resulting in higher usage fees and rsi on my thumb for scrolling 90% of messages including all general email conversations are too posted yet discussion groups want bottom posting. Why? This is a different argument here. The problem is that all to often people quote irrelevant text. Now, if someone had just read my original top reply he could have concluded that you have no idea why top-posting is about and need to be tought the basics. This is because I have replied to your message outside of context. What else have you quoted: [ Snip 11 lines of signature ] - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Sent: Thu Jun 21 12:48:45 2007 Subject: Re: [asterisk-users] Asterisk GUI Four lines of headers (the bad headers quoting style) that actually leave out the name of the poster. On Jun 20, 2007, at 5:04 PM, Troy Ayers wrote: I would have been convinced if you had not top-posted! heh The actual text you replied to. 24 lines of unrelated text from the original message Tom left in for instructive purposes and you have not bothered trimming: Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob OK, Now I'm confused... I was prepared to accept Rob's argument due its beautiful, flawless logic. But Troy has a valid point: Rob did top-post, invalidating his point. But so did Troy, invalidating his point, so now I'm stuck. Whatever shall I do? I think I'll just stick with my own opinion, seeing as both Rob and Troy are obviously idiots. (duh!) ;-) Tom 7 lines of of mailing list footer. Tom has trimmed the unnecessary ones there (and I have removed the extra one added by the mailing list manager to your message when it got to the list. ___ --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 So instead of complaining about others who force you to scroll, trim the useless stuff. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind xfer issue -- URGENT!
That's exactly what is happening. The *caller* is hitting #0 and
transferring the *agent* (my operator) to the new number. I don't have
the 'T' flag set [exten = s,n,Queue(queue-answer|t|||20)], so I was led
to assume that the caller could not transfer. Am I wrong?
Jay
Wes Baehr wrote:
It sounds more like the agents are making the transfers...
If a caller were to transfer a call (#0 1555-555-1212), they would be
transferring the AGENT to the that number, not themselves!
Either way, the caller SHOULD be disconnected after the transfer. (Or
perhaps leaked somewhere else into the dialplan they shouldn't be going,
which lets them dial out long-distance.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, June 21, 2007 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind xfer issue -- URGENT!
Use the dialplan show CLI command (show dialplan in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be
the first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find. There must be leak,
check if you are using any wrong extension patterns like _XXX. or
something like that.
On 6/19/07, *Jay Moore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The way I have my dialplan set up, the callers shouldn't be able to
make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file
and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing
everything
properly. Callers should not be able to transfer (no 'T' in the
Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an
idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only
your local
office extensions.
I assume all your incoming calls fall in [default] context. what
you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office
extensions.
I hope you get my point
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com http://www.axvoice.com
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Re: [asterisk-users] Query
On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh Either contact digium support or post the problem ram ___ --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] Once Touch Recording
Klaverstyn, David C wrote: Hi All, Once touch recording only seems to work between extensions. When calling an external party when pressing *1 does nothing. The person you have called can hear 2 DTMF tones. Is there a trick to getting once touch recording working over a zap channel? I am using a TE110P, but calls over SIP to a VSP also fails when trying to use one touch recording. The trick we use is to include W in the Dial() options. :-} Have you checked the Dial() command for external calls? regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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] chan_features.so / asterisk 1.4.5
In my asterisk 1.4.5 chan_features.so has been installed properly... check in your asterisk-source if /channels/chan_features.so is present regards, Jack escribió: Hi, after updating from asterisk 1.4.4 to 1.4.5 I get a warning for chan_features.so: Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. Is chan_features.so deprecated for asterisk 1.4.5 or why is this module not installed by asterisk 1.4.5? Regards, Jack ___ --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 -- José Luis Ledesma Tecnobe Tecnología S.L. ___ --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 0 dial outgoing call
On Fri, 2007-06-22 at 05:59 -0700, satish patel wrote: can u give me example how do i create plan for this task or job ram [EMAIL PROTECTED] wrote: On 6/22/07, satish patel [EMAIL PROTECTED] wrote: Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call create dialplan for the same ram What digit do you dial on the avaya to get PSTN dialtone? Setup a dial plan entry for dial digit 0 to access the avaya and dial the access code for the PSTN . dave ___ --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] Query
ram wrote: On 6/22/07, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, Can anybody tell me that wether I should install DIGIUM-TE120P card on redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of modutils and iptable. Can anybody help me out of this. Thanx and Regards sanchal singh Either contact digium support or post the problem ram Time for CentOS. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk - no audio
Hi Demuel, On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yeah, just the same as the sample configuration by mog. However, if I am using a gtalk application in asterisk to dial googletalk buddy, my voip phone is suddenly connected to the googletalk buddy though at the googletalk client software it is still waiting to be accepted or not accepted. I mean from my voip phone perspective, there is just one ring to make a call to the googletalk buddy unlike in the jingle application wherein there are successive ring before the googletalk buddy accepts the call. That's strange. I was not able to reproduce this problem, that is, when dialing an extension that points to a GoogleTalk client from a SIP phone, I *always* have to wait for the GoogleTalk client to accept the call. That's just like if you had Asterisk automatically answer GoogleTalk calls. Do you have any file streamed to the SIP phone by Asterisk? Philippe ___ --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] chan_zap problems
Hi, I have Asterisk 1.4.0 using Queue App. I use PRI connecting my Asterisk with Siemens EWSD This was working OK but since two days I have this error: [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:14] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:15] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:15] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. I the EWSD try to send a call in this channel I hear a busy tone. In the others channels asterisk answer ok. And when it happends only restarting Asterisk I can fix the error, it happends two times a day. 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
Re: [asterisk-users] got-name
Bill Michaelson wrote: Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? I don't know how to contact them, but I am having the same problem. ___ --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] different codec for different extensions
Hi, what about this: when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec exten = 111,1,Set(SIP_CODEC=gsm) exten = 111,2,Dial(SIP/.) When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec exten = 222,1,Set(SIP_CODEC=alaw) exten = 222,2,Dial(SIP/.) Nasir Iqbal wrote: Hi Mojo, I dont have control our calling party. and also called extension is only configured in extensions.conf not sip.conf etc. So I must select the codec within my dialplan (extensions.com) I found one solution by using SIP_CODEC variable like [fax] exten = 605,1,ringing() exten = 605,n,set(SIP_CODEC=ulaw) exten = 605,n,RxFAX(/tmp/nasir.tiff|ecm) exten = 605,n,hangup() but Thanks for your answer Thanks Nasir Iqbal [userX] ... context=internal disallow=all allow=gsm allow=ulaw ... [fax] ... disallow=all allow=ulaw ... Then any IVRs that userX accesses should be in gsm because it's the preferred codec? Assuming that the gsm sound files ARE installed? You might experiment with this. But when userX is bridged to the fax channel, ulaw is the only one the fax channel allows, so it's chosen on both ends. Shouldn't this work? Mojo Nasir Iqbal wrote: Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --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 -- thanks, Yusuf ___ --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] PhpAgi call generation
Thanks Lee,
That really helped me to get my project started... I am in process of
developing IVR based Notification System which is going to integrate
with my IVR based Time clock system.
Notifications will be based on, if an employee is late to clock in,
event should trigger and generate a .call file and call the supervisor
and let him know that XYZ employee is late, do you want to inform an
employee... etc...
Cheers,
Nitesh
Lee Jenkins wrote:
Nitesh Divecha wrote:
Is there any info on how to create .call files with some examples? And
where to place this file? And how to initiate it..?
Thanks man...
Cheers,
Nitesh
Christopher Robinson wrote:
That should be pretty easy to do with a .call file. The context that
you drop your called party off to will play the sounds and do the
transfer. So really you need to concentrate on creating that context,
the .call files are very easy to generate.
Nitesh Divecha wrote:
Finally, this is what I was looking for... to generate a call.
I have been working on my Time Clock application, where an employee will
call into the system using his cellphone to clock in and clock out his
hours. And it works perfect...
Now I was looking for an option where or if an employee is late to clock
in, the system has to generate a call and call the supervisor and inform
him that XYZ employee is late and give an option to supervisor Would
you like to call XYZ employee, Press 1 and the system will call the XYZ
employee and connect with the supervisor...
Is it something feasible to do using the .call files? Or I am way too
off...
Cheers,
Nitesh
Christopher Robinson wrote:
I've done this many times, also used the .call files. If you don't need
your application to initiate the call the .call files are the better way
to go, otherwise it's a bit too much file management overhead.
Here's some working code on our end. In this case the Channel is
actually a context which makes the actual call, but I've used it both
ways.
?php
require('PHPAGI/phpagi-asmanager.php');
$callid = 'Somebody';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=LOCAL/[EMAIL PROTECTED],
'Context'='called_party_context',
'Exten'='899',
'Timeout' = '1000',
'Async'='1',
'MaxRetries' = '5',
'RetryTime' = '5',
'Priority'=1,
'Callerid'=$callid));
$asm-disconnect();
}
?
nik600 wrote:
hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
?php
$asm = new AGI_AsteriskManager();
if ($asm-connect()) {
$asm-Originate(Zap/g1/1,number,default,1);
/*
play message...
*/
} else {
die(error\n);
}
?
But it doesn't work.
Is it possible to create a program like this?
thanks
Sorry, I can't help you with PHP. All my stuff is in pascal. But here
is a link to call origination info:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
I did something a bit like what you're doing, but it was a script to
call into the system and generate a broadcast type message to a
different party. Again, a bit different, but the elements are all the
same; call control, origination, database access, etc. Its in pascal,
but the syntax is very easy to understand and may give you an idea of
how program flow might be.
http://www.leebo.dreamhosters.com/apscripts/msgcast/
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Re: [asterisk-users] POTS - Incoming Voice or Fax - How to tell?
On Fri, 22 Jun 2007, Gary wrote: Basically, how does the machine know if the incoming call is a fax or voice call? It quickly listens for fax tones - certain sequences of detection tones from the other end that are in a particular acoustic band. Zaptel supports this on its interfaces (including FXO/POTS), although I have not tried it personally and do not know how well it works: http://www.voip-info.org/wiki-Asterisk+fax#Zapfaxdetection -- Alex -- 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] got-name
Daryl Jones wrote: Bill Michaelson wrote: Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? I don't know how to contact them, but I am having the same problem. Is this who you mean? http://got-name.com/contact.php Got Name, Inc. 12345 Lake City Way NE Seattle, WA 98125 Phone: 1-727-254-4000 Email: [EMAIL PROTECTED] The domain is registered to Jed Stafford. If you want the domain contact details you can do a whois. I won't post them here since I'm sure they don't want even more spam. ___ --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] international numbers...
Try 00 as a sub for the + in the search. That's how the chan_skype dials it so possibly your dial range becomes: 0061|0+. on the outgoing route. Just guessing Let me know if it works D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri Jun 22 05:10:46 2007 Subject: [asterisk-users] international numbers... Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +61242110 to something like 02422110 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XX but that still seems wrong to me. it could match other numbers. anyone had to do this in the past ? 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
[asterisk-users] searching for compatible servers
Im trying to find the best hardware to run asterisk on. I see that the compatibility list is a little dated. Any recommendations out there? This is for a 19 phone system with 2 tdm cards… Thanks Hart Green -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/705 - Release Date: 2/27/2007 3:24 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to config SIP blind transfer in extension.conf
You can find detailed info about command Transfer at http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer . _ From: Lucian Romi [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 19, 2007 2:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to config SIP blind transfer in extension.conf I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten = 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know the syntax for that. I tried exten = 511,1,Transfer(SIP/sip_proxy-out) So can any one give me a hint on this? 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
Re: [asterisk-users] problem with one way audio
Don Briggs wrote: I have a company with asterisk 1.2.19 and polycom 501 phones. I get one way audio. A caller from the pstn world hits the tdm400 card, This rings two phones in a ring group. My client answers the phone, the calling party is told the customer here her but she can not here them. The customer hangs up and calls back and the call goes through.. I rolled back to 1.2.14 and the problem is much better but is still there, Are there any ideas Don Briggs 573-614-5667 ext 4037 Do you have CallProgress=yes in your zapata.conf? This one just bit me in the arse this morning. I set it to no and one-way audio went away. -- 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
[asterisk-users] Hints
I used to run Asterisk 1.4.4 but had to revert back to 1.2.13 to minimize a bug we were coming across. 1.4.5 looked promising, but the hints are broken and making it so I'll likely have to go back to 1.2.13 until I get the hints fixed. I'm using Grandstream phones hints on the parked extensions. I should also clarify that when I upgraded versions, I renamed all Asterisk folders (/var/log /var/lib /usr/lib /var/spool) so I could have a 'clean' install of 1.4.5. There's a few things that are happening on hints. First, on a fresh reboot, despite the server saying all hints are IDLE the Grandstream phones light up as if INUSE. This has never happened across umpteen different versions of Asterisk I've ran. The fix is to actually put the parked extensions INUSE and clear them, then they function fine...for a bit. The second problem is, after about an hour, hints just stop working. Well, hints actually work, but the phones stop watching. [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 [EMAIL PROTECTED] : park:[EMAIL PROTECTED] State:IdleWatchers 29 The watchers in an hour or so after a fresh reboot will drop to 0, I believe it has to do with when the phone reregisters. Which brings me to the third problem (directly related to a phone reregistering). After a fresh reboot, if I reboot all phones before any calls get parked, all phones work properly (for an hour anyway). However, if I reboot a phone *after* a calls been placed on hold, the hints do not work for that phone and the Watchers doesn't get updated (say I have Watches:28 and I plug another phone in, it should go to 29 but it won't unless I restart Asterisk). So somewhere I've got something messed up. Not sure where to look, it seems odd that as soon as the parking lot is used (and a hint updated) it kills any new watchers from attaching, as well as all watchers drop off after an hour. Any thoughts on where to look? ___ --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 GUI
Dave Bour wrote: So I'll ask the question. What's wrong with top posting. WOW! Is this a mine field, or what? You have stumbled into one of the hot religious arguments on just about all lists. There will NEVER be an agreement on which is acceptable. Many anchored in the past hotly content that one be drawn and quartered, or at the least banished to Gitmo for top posting. Some of these same people don't ever bother to trim the tag lines found on every posting, so one has to wade through several tag lines to even find if someone posted or simply had a twitchy finger. I am sure it HAS to be even worse on a Blackberry. The good news with Blackberries is that you won't have the French ( government ) folks clogging up the works. Incurable Top Poster. ___ --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] Binding to multiple ports in sip.conf
I'd like asterisk to bind to multiple ports in sip.conf. Is this possible? Something like bindport=5060, Thanks Suresh ___ --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] gtalk - no audio
Hi Philippe, In my /etc/asterisk/extensions.conf, I tested both Gtalk/asterisk/googletalkbuddy and Jingle/asterisk/googletalkbuddy. When using Gtalk/asterisk/googletalkbuddy, it is consistent with making a call to googletalk buddy but it just ring once. After the ringing, it just displayed on the voip phone that it is connected to the googletalk buddy and the timer clock in its lcd starts incrementing. Though at the googletalk buddy, it just indicates that somebody is calling and waiting to be answered. But when accepting the call, I still got audio both ways. Now, when using Jingle/asterisk/googletalkbuddy, I got a ring until such that the googletalk buddy accepts the incoming call then the ringing stops but I could not hear any audio at all. FYI, I don't have any problem with making a call from the googletalk client to asterisk. What is the main distinction between Jingle and Gtalk here? How should I generate the file streamed to the SIP phone by Asterisk? Regards, Demuel Hi Demuel, On 6/22/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Yeah, just the same as the sample configuration by mog. However, if I am using a gtalk application in asterisk to dial googletalk buddy, my voip phone is suddenly connected to the googletalk buddy though at the googletalk client software it is still waiting to be accepted or not accepted. I mean from my voip phone perspective, there is just one ring to make a call to the googletalk buddy unlike in the jingle application wherein there are successive ring before the googletalk buddy accepts the call. That's strange. I was not able to reproduce this problem, that is, when dialing an extension that points to a GoogleTalk client from a SIP phone, I *always* have to wait for the GoogleTalk client to accept the call. That's just like if you had Asterisk automatically answer GoogleTalk calls. Do you have any file streamed to the SIP phone by Asterisk? Philippe ___ --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] access to asterisk server since internet
hi i have configured an asterisk server which i have tested locally with x-lite and that's ok but when i wanted to access to it since internet that hasent taken place knowing that my server has access to internet by a wifi router that has a public ip address (e-g a.b.d.c) and asterisk server has a private ip 192.168.1.111 (the firewall is disabled) can some one tellme how i should configure the x-lite clients with this configuration and what should i change to access to my server since internet -- Scales of success are not easy to be ridden ___ --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] got-name
I don't know how to contact them, but I am having the same problem. The domain is registered to Jed Stafford. If you want the domain contact details you can do a whois. The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives. --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does Early Media have to be Ulaw?
I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid=Test hone 1 +19256002182 host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default insecure=port dtmfmode=rfc2833 canreinvite=yes qualify=yes disallow=all ;allow=ulaw allow=g729 Level 3 sends early media... --- Transmitting (no NAT) to xxx.yyy.34.210:5061 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.yyy.34.210:5061;branch=z9hG4bK-tenor-d802-22d2-004d;received=xxx.yyy .34.210 From: sip:[EMAIL PROTECTED];tag=d80222d2-27 To: sip:[EMAIL PROTECTED];tag=as4fe079a5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] ontent-Type: application/sdp Content-Length: 261 v=0 o=root 2235 2235 IN IP4 xxx.yyy.34.195 s=session c=IN IP4 xxx.yyy.34.195 t=0 0 m=audio 10484 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv and Asterisk responds on the console with: [Jun 22 10:06:03] WARNING[32573]: channel.c:2882 set_format: Unable to find a codec translation path from g729 to slin [Jun 22 10:06:03] WARNING[32573]: indications.c:121 playtones_alloc: Unable to set 'SIP/19256002182-096ac918' to signed linear format (write) This doesn't happen when progressinband=no. It almost seems like Asterisk has to do early media as G711 only. Is that the case??? Doug. ___ --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] Blind xfer issue -- URGENT!
Just out of curiosity, could you 'show queues'?
Thanks.
Wes Baehr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Friday, June 22, 2007 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind xfer issue -- URGENT!
That's exactly what is happening. The *caller* is hitting #0 and
transferring the *agent* (my operator) to the new number. I don't have
the 'T' flag set [exten = s,n,Queue(queue-answer|t|||20)], so I was led
to assume that the caller could not transfer. Am I wrong?
Jay
Wes Baehr wrote:
It sounds more like the agents are making the transfers...
If a caller were to transfer a call (#0 1555-555-1212), they would be
transferring the AGENT to the that number, not themselves!
Either way, the caller SHOULD be disconnected after the transfer. (Or
perhaps leaked somewhere else into the dialplan they shouldn't be going,
which lets them dial out long-distance.)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, June 21, 2007 6:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Blind xfer issue -- URGENT!
Use the dialplan show CLI command (show dialplan in 1.2) to show
you exactly what asterisk has picked up, and scan it for aforementioned
leaks.
Rizwan Hisham wrote:
Then i think u should use Atis's idea of using transfer_context
variable...you should set it inside your dialplan and it should be
the first thing you do in your dialplan.
Are you sure there is no leak in your dialplan, because asterisk cant
transfer your caller to an extension it cant find. There must be leak,
check if you are using any wrong extension patterns like _XXX. or
something like that.
On 6/19/07, *Jay Moore* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
The way I have my dialplan set up, the callers shouldn't be able to
make
any outgoing calls.
Incoming calls come down my T1:
{zapata.conf}
; T1
group=1
context=incoming_t1
signalling=em_w
channel = 1-24
Which puts them into the 'incoming_t1' context:
{extensions.conf}
[incoming_t1]
#include callcenter/extension_ans.conf
include = answering-service
Which includes my callcenter answering service extensions conf file
and
includes the 'answering-service' context:
{callcenter/extension_ans.conf}
[answering-service]
; Catch all extensions
exten = _X.,1,Set(account=${EXTEN})
exten = _X.,n,AGI(get_cid.php)
exten = _X.,n,Set(CALLERID(all)=${cid}${account})
exten = _X.,n,Set(context=COM)
exten = _X.,n,Set(type=INC)
exten = _X.,n,Set(from=${account})
exten = _X.,n,Set(to=COM)
exten = _X.,n,AGI(create_filename.php)
exten = _X.,n,Set(MONITOR_FILENAME=${filename})
exten = _X.,n,Goto(queue-answer,s,1)
Which then parses all incoming calls (you can see the rest of the
dialplan in my previous message).
I'm not sure what I'm doing wrong. It seems to me I'm doing
everything
properly. Callers should not be able to transfer (no 'T' in the
Queue()
command), and they should not be able to dial any extension.
I'm completely lost here.
Jay
Rizwan Hisham wrote:
I dont know how to solve your transfer problem, but i have an
idea which
you
can use to overcome this abnormality.
You should restrict the callers with context which includes only
your local
office extensions.
I assume all your incoming calls fall in [default] context. what
you should
do is:
[default]
include= localext
exten= _X.,1,Noop(Incoming call received)
[localext]
*This context should include all your office extensions**
This way, caller can only transfer himself within your office
extensions.
I hope you get my point
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
www.axvoice.com http://www.axvoice.com
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[asterisk-users] Audio going one way for a few seconds during the call
Hi, This question was posted earlier, but there was no satisfactory answer to it. Afterwards I tried everything but to no avail. The problem of audio going one way during the call for a few seconds is still there. Its Asterisk 1.2.18 hosted Dell server with no NAT. Phones connect remotely through a hi-speed Internet connection, they are behind NAT on a D-Link router, UDP ports 5060, 10001-2 are forwarded to LAN,*, which means they are forwarded to all the IPs. How can I fix this problem. -- Zeeshan A Zakaria ___ --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] got-name
Luki wrote: I don't know how to contact them, but I am having the same problem. The domain is registered to Jed Stafford. If you want the domain contact details you can do a whois. The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives. It looks like they're in the same office. Also SellVoip.net is a trade name of Blue Networks, Inc., and it looks like Jed has lost his corporate registration. See: http://www.secstate.wa.gov/corps/search_detail.aspx?ubi=602464110 This usually isn't a big deal. In most, if not all, states he can just pay a penalty and get it reinstated. However, I can't find any record of Got Name, Inc. in the State of Washington. This is supposed to be the business entity under which got-name.com is operating, and that is a big deal. --Jeff ___ --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] Ring/Off-hook in strange state 6
HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --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] searching for compatible servers
Everyone is going to have their sacred cow on this one so suspect you might have opened a can of worms ... I can tell you that I have very good results using a number of different Intel based SuperMicro servers ... these seem to be very mundane and extremely well behaved ... I have used both Digium and Sangoma cards in them (TDM only, have not tried T1's or ISDN) ... my only beef with them is that they seem rather noisy (very loud cooling fans) ... I have also used a couple entry level Intel based Dell servers with good results and can tell you that these seem to be a good bit quieter than the SuperMicro ... however, the quality of construction and components used on the Dell seems inferior to the SuperMicro ... I have also used a couple mid range HP servers with good results ... the HP is very nicely made and seems to be a notch above the SuperMicro in terms of overall quality of construction and components used ... however, they are about 20% more expensive in similar configuration ... I have had good results using the new 300mb SATA Raid setup from Adaptec ... I normally use CentOS as my OS and the installation utility finds the controller and could not be any simpler ... would expect similar with most RedHat based Linux flavors ... in general, have always had good luck with Adaptec drive controllers ... just be careful to use SATA drives that are specifically intended for use in a RAID, not common workstation drives ... there is a difference and it can bite you in the hind quarters if you buy the wrong type of hard drives and try to use them in a RAID ... Did recently have some trouble with an Intel 1gb NIC ... this surprised me ... I have always favored Intel NIC's mainly because I am lazy and the OS just seemed to find them without having to jump through any hoops ... but this fancy new server class 1gb Intel NIC required that I hunt down and install a unique driver for a CentOS 4.x install ... but this was an odd ball ... most 10/100 and older 1gb Intel NIC's have worked without issue for me ... have had generally good experience with 3Com and Realtec also ... I think the only server class hardware that I recall giving me fits was an ancient Compaq server that someone gave me ... I messed with that one for a week or so on and off and never did get the darn thing to run Linux let alone Asterisk ... As far as I can tell, the only really temperamental aspect is TDM cards from Digium ... while the cards are generally of decent quality, they seem to be a bit picky about what kind of PCI slot they will work with ... so far, this has not been a major problem for me as the hardware I used is purposely very mundane ... but with the published compatibility list hopelessly out of date, you stand some risk of buying a server with a motherboard that the Digium TDM card will not take to ... I have NEVER heard of this problem with Sangoma cards ... Most of my installs these days are on embedded hardware ... I favor the Astlinux flavor of Asterisk and like my PBX's to be small, fanless, lean and mean ... for these I have tried a number of fanless type barebones systems and finally settled on the Lex Neo/Twister models as being my production standard ... these are VIA C3 1ghz machines that are similar to a Mini-Itx ... the Lex Twister model will handle a Digium TDM card nicely and still have room for a 2.5 in hard drive if you want one ... the Lex Neo has no card slot so is not suggested if you want a PCI card that supports connection to the PSTN, but will take a 2.5 in hard drive ... both models have 3+ Realtec NIC's built in which works well with Astlinux when used in router/firewall mode ... With Astlinux, I normally boot off a CF card and forego the moving parts associated with the hard drive but to each his own ... anyway, them's my 2 cents ... Regards G.Hendershot From: Hart Green [mailto:[EMAIL PROTECTED] Sent: Friday, June 22, 2007 11:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] searching for compatible servers Im trying to find the best hardware to run asterisk on. I see that the compatibility list is a little dated. Any recommendations out there? This is for a 19 phone system with 2 tdm cards. Thanks Hart Green -- Internal Virus Database is out-of-date. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.4/705 - Release Date: 2/27/2007 3:24 PM ___ --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] got-name
On Fri, 22 Jun 2007, Luki wrote: I don't know how to contact them, but I am having the same problem. The domain is registered to Jed Stafford. If you want the domain contact details you can do a whois. The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives. --Luki Oh, you're kidding me!? Oh geez. Guess that's *another* lesson to learn. Always check the whois on a domain and compare it against Google searches for complaints before you do business with a new company. I guess I can kiss my $5.00 goodbye. Luckily it was only $5 and I didn't pay for more yet. What bothers me more than losing the $5 is the fact that I STILL need a CNAM service, and I don't want to pay the huge amount my CLEC wants for it. Anyone know of any other CNAM services, preferably NOT run by this guy? -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] inband DTMF for g729
Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nuance Buys Tegic from AOL for $265m
Nuance Communications has agreed to buy Tegic Communications, the developer of the T9 predictive text input software for mobile phones, from AOL for $265 million in cash. http://www.wirelessweek.com/article.aspx?id=149702 Article goes on to say T9 is in use on over 2.5billion phones - wow now that's a patent worth filing. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] international numbers...
KW == Kevin Withnall [EMAIL PROTECTED] writes:
KW Using trixbox (or a custom dialplan if needed) has anyone been
KW able to convert a number dialled like +61242110 to something
KW like 02422110 ie (remove the +61 and replace with 0)
KW i just dont know how to set it up, there seems to be no dialplan
KW wildcard i can use to match +.
The easy way out has served me well in the past. Something like:
_+61!,1,Goto(0${EXTEN:3},2)
_+!,1,Goto(00${EXTEN},2)
_X.,1,NoOp
_X.,2,...
Notice that the extension reordering that asterisk does can easily
mess you up. It's important to do show dialplan afterwards, to see
what asterisk came up with this time. The above is untested.
/Benny
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[asterisk-users] H.323 IP Phones and H.323 Traffic
Hi List; I saw sip.conf and iax.conf but I do not see a files for H.323 IP Phones, does that mean Asterisk does not support H.323 IP Phones? Also, what if Asterisk need to talk with another IP PBX that support H.323, so the IP Trunk in that case should be H.323 IP Trunk, does Asterisk support such thing? Regards Bilal Ghayad Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Binding to multiple addresses
I have a simliar problem as the port binding question. I have a four port parelell processing NIC, I would like to team them together. Can I do this in asterisk if they are not actually teamed in hardware. I would be binding to several addresses simultaniously. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.5
I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED] Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday June [EMAIL PROTECTED]:30PM EDT Asterisk Users Conference
On 6/22/07, randulo [EMAIL PROTECTED] wrote: Quick reminder that the conference is happening today at 12:30 PM EDT. Listen to the conference here: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 The little round orange Listen button will open a player. You can also just download an mp3 via the Download button or here: http://recordings.talkshoe.com/TC-22622/TS-26149.mp3 Bryan had some interesting stuff for us. I'm trying to organize a video/audio simulcast with Mark Spencer in two weeks, watch for it on July 6th. If you want to be a guest in the coming weeks, don't hesitate to contact me off list. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got-name
I started doing HTTP queries with curl from my own AGI script and that
still works.
Their example doesn't work.
You can add this to the callerid_shell.agi script floating around.
lookup_gotname() {
out=
out=`/usr/bin/curl -s -m 2 -A Mozilla/4.0
http://cnam.got-name.com/?auth=USERNAME:PASSWORD\type=http\number=${1}`
echo $out;
}
-Jonathan
Bill Michaelson wrote:
Is it just me, or is the AGI interface at cnam.got-name.com failing
for others? Anyone know how to contact them without sending postal
mail or telegram?
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Re: [asterisk-users] inband DTMF for g729
I have a similar issue with Qwest SIP. They only support rfc2833 in g729AB, and Asterisk is only G729A. Sprint works fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Fredrickson Sent: Friday, June 22, 2007 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inband DTMF for g729 Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
We are using Level 3. At this point, changing carrier is not an option. - Original Message - From: Matthew Fredrickson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 22, 2007 3:20 PM Subject: Re: [asterisk-users] inband DTMF for g729 Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ex-Girlfriend Logic in 1.4.4
Mojo with Horan Company, LLC wrote:
For real? I thought _ was to tell asterisk it was time for some pattern
matching:
; exact extension, exact cid
exten = 5000/19256002182,1,Answer
; any extension beginning with 5, from specific cid only
exten = _5./19256002182,1,Answer
; match exactly extension 5000, but anyone calling from
; (925) 600- matches
exten = 5000/_1925600.,1,Answer
; match anyone calling any extension beginning with 5 FROM any cid
; in the (925) 600- block
exten = _5./_1925600.,1,Answer
are the ways I've always used the underscore.
Doug, sorry I didn't have anything to help with your problem. I just
wanted to get some clarification of this poster's statement, to either
help myself or 10,000 other readers, I'm not sure who, yet...
Mojo
Nasir Iqbal wrote:
Hi,
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the ca
Try with underscore before extension like.
exten = _5000/19256002182,1,Answer
Nasir Iqbal
ICT Innovations
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The only problem I see is that you have a 1 in the number, check the
console when that call comes in, it probably doesn't have the 1 in it as
most CID does not include the country code.
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Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 81
Yes, of course. What happens when you dial the number, Daryl? Daryl Jones wrote: Bill Michaelson wrote: Is it just me, or is the AGI interface at cnam.got-name.com failing for others? Anyone know how to contact them without sending postal mail or telegram? I don't know how to contact them, but I am having the same problem. Is this who you mean? http://got-name.com/contact.php Got Name, Inc. 12345 Lake City Way NE Seattle, WA 98125 Phone: 1-727-254-4000 Email: [EMAIL PROTECTED] smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inband DTMF for g729
On 6/22/07, Gary Chen [EMAIL PROTECTED] wrote: We are using Level 3. At this point, changing carrier is not an option. Gary, I use Level(3) with G729a and RFC2833. No problems, no requirement for inband G729. -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ring/Off-hook in strange state 6
Alex, I had this problem with a new TDM2400 card that we purchased. Specifically I would get that message and then it would pick up the ringing line AND the line next to it. Basically, lines 1 2 had been cross-linked somehow. After a few weeks of trouble-shooting with Digium tech support they cross-shipped me a new card and the problem (and that message) went away. Daniel Hazelbaker High Desert Church On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote: HI I have two servers both of which get this message on one of the lines. Ring/Off-hook in strange state 6. The one server seems to be ok with it, but the other one when an extension picks up there is no one there and the incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like someone had suggested, but it didn't do anything. I also upgraded zaptel to the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to no, as well as busydetect=no. This is a major problem since this box only has 1 other line, but at least it works. I can't seem to find much info on this issue. I can't believe others haven't run into it. I started a ticket with digium, but I guess they are pretty backed up. Here is what I am getting in the CLI: Thanks for any help -Alex -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 is ringing Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 4 -- SIP/4125-09559118 answered Zap/4-1 == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1' -- Hungup 'Zap/4-1' -- Starting simple switch on 'Zap/4-1' -- Executing Wait(Zap/4-1, 1) in new stack -- Executing Answer(Zap/4-1, ) in new stack -- Executing Dial(Zap/4-1, SIP/4125|20|tr) in new stack -- Called 4125 -- SIP/4125-09559118 is ringing ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 301 - Problem with AMI Originated Calls
Hi all,
I'm having an odd problem with my polycom 301. I am initiating a call
to it with AMI Originate() function:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: dropped_conf=111
The to_meetme context is very simple:
[to_meetme]
exten=s,1,MeetMe(${dropped_conf},id)
If I specify every other device I have to test:
* Grandstream 101
* XLite Client
* My Cell Phone
It works as expected. But with the Polycom, the phone will ring and the
usual ANSWER REJECT FORWARD soft buttons are painted on the display, but
hitting the answer button seems to fail to do anything other than
silence ringing.
SHOW CHANNELS shows the polycom as ringing still although the polycom
has stopped ringing (audibly at least).
Of course, all other calls originate through the dialplan are answered
with no problem.
Anyone have an idea what might be causing this? Its a polycom 301 with
lines 1 2 registered to separate sip accounts in sip.conf.
Thanks for any suggestions.
--
Warm Regards,
Lee
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Re: [asterisk-users] gtalk - no audio
What is the main distinction between Jingle and Gtalk here? How should I generate the file streamed to the SIP phone by Asterisk? I really have no clue :). Maybe you can open a bug report so that we can dig into this problem. Thanks! Philippe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
John Novack wrote: Dave Bour wrote: So I'll ask the question. What's wrong with top posting. WOW! Is this a mine field, or what? You have stumbled into one of the hot religious arguments on just about all lists. There will NEVER be an agreement on which is acceptable. Many anchored in the past hotly content that one be drawn and quartered, or at the least banished to Gitmo for top posting. Some of these same people don't ever bother to trim the tag lines found on every posting, so one has to wade through several tag lines to even find if someone posted or simply had a twitchy finger. I am sure it HAS to be even worse on a Blackberry. The good news with Blackberries is that you won't have the French ( government ) folks clogging up the works. Incurable Top Poster. ___ --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 Hi all! Late pointless post type o' troll here to say wow has this gone way off OOT. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STDERR in AGI
Ronaldo Z. Afonso wrote:
Hi all,
I just started programming using AGI and I have a simple question about
STDERR.
If I understood it right, all the messages sent to STDERR should be
shown in the Asterisk console, but using the following python code I
just can't see anything.
#!/usr/bin/python
#
# File: /var/lig/asterisk/agi-bin/agi-test.py
#
# Description: An AGI Script
#
import sys
env = {}
tests = 0
while True:
line = sys.stdin.readline().strip()
if line == '':
break
key,data = line.split(':')
if key[:4] != 'agi_':
sys.stderr.write(Did not work!\n)
sys.stderr.flush()
continue
key = key.strip()
data = data.strip()
if key != '':
env[key] = data
sys.stderr.write(AGI Environment Dump:\n)
for key in env.keys():
sys.stderr.write( -- %s = %s\n % (key,env[key]))
sys.stderr.flush()
##
This code comes from the book Asterisk: The future of the Internet and
it is being activated by an extension like that:
exten = 123,1,Answer()
exten = 123,2,AGI(agi-test.py)
Any help would be appreciated.
Ronaldo.
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STDERR goes to console.
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Re: [asterisk-users] PhpAgi call generation
Nitesh Divecha wrote:
Thanks Lee,
That really helped me to get my project started... I am in process of
developing IVR based Notification System which is going to integrate
with my IVR based Time clock system.
Notifications will be based on, if an employee is late to clock in,
event should trigger and generate a .call file and call the supervisor
and let him know that XYZ employee is late, do you want to inform an
employee... etc...
Cheers,
Nitesh
Lee Jenkins wrote:
Nitesh Divecha wrote:
Is there any info on how to create .call files with some examples? And
where to place this file? And how to initiate it..?
Thanks man...
Cheers,
Nitesh
Christopher Robinson wrote:
That should be pretty easy to do with a .call file. The context that
you drop your called party off to will play the sounds and do the
transfer. So really you need to concentrate on creating that context,
the .call files are very easy to generate.
Nitesh Divecha wrote:
Finally, this is what I was looking for... to generate a call.
I have been working on my Time Clock application, where an employee will
call into the system using his cellphone to clock in and clock out his
hours. And it works perfect...
Now I was looking for an option where or if an employee is late to clock
in, the system has to generate a call and call the supervisor and inform
him that XYZ employee is late and give an option to supervisor Would
you like to call XYZ employee, Press 1 and the system will call the XYZ
employee and connect with the supervisor...
Is it something feasible to do using the .call files? Or I am way too
off...
Cheers,
Nitesh
Christopher Robinson wrote:
I've done this many times, also used the .call files. If you don't need
your application to initiate the call the .call files are the better way
to go, otherwise it's a bit too much file management overhead.
Here's some working code on our end. In this case the Channel is
actually a context which makes the actual call, but I've used it both
ways.
?php
require('PHPAGI/phpagi-asmanager.php');
$callid = 'Somebody';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=LOCAL/[EMAIL PROTECTED],
'Context'='called_party_context',
'Exten'='899',
'Timeout' = '1000',
'Async'='1',
'MaxRetries' = '5',
'RetryTime' = '5',
'Priority'=1,
'Callerid'=$callid));
$asm-disconnect();
}
?
nik600 wrote:
hi
i'd like to write a simply application in php with phpAgi that:
- connect to Asterisk
- call an external number using a Zap channel
- play a message
here is some code:
?php
$asm = new AGI_AsteriskManager();
if ($asm-connect()) {
$asm-Originate(Zap/g1/1,number,default,1);
/*
play message...
*/
} else {
die(error\n);
}
?
But it doesn't work.
Is it possible to create a program like this?
thanks
Sorry, I can't help you with PHP. All my stuff is in pascal. But here
is a link to call origination info:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
I did something a bit like what you're doing, but it was a script to
call into the system and generate a broadcast type message to a
different party. Again, a bit different, but the elements are all the
same; call control, origination, database access, etc. Its in pascal,
but the syntax is very easy to understand and may give you an idea of
how program flow might be.
http://www.leebo.dreamhosters.com/apscripts/msgcast/
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It should be noted that the above code isn't AGI. its using AMI, the
Asterisk Manager Interface.
An example of how to use it is:
$message = Action: Originate\r\n
Channel: Local/3035551212\r\n
MaxRetries: 5\r\n
RetryTime: 300\r\n
WaitTime: 45\r\n
Context: redalert\r\n
Exten: s\r\n
Priority: 1\r\n
Callerid: 1234\r\n\r\n;
$response=$astman-ast_cli(localhost, $message);
Of course this is using my own custom AMI PHP module, but the elements
needed to setup a call are the same. This particular call plays a
recorded message after the phone is answered.
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Re: [asterisk-users] Query
The best person to check with is Digium support. They have support matrix for
Kernel hardware on which ur card will perform.
Please check the compatibility matrix. Should work fine with
http://www.digium.com/en/supportcenter/documentation/viewdocs/TE120P
Digium support. 256-428-6000
[EMAIL PROTECTED] wrote: Hi all,
Can anybody tell me that wether I should install DIGIUM-TE120P card on
redhat9.0 2.4.20-8 or 2.6.18. I am using kernel 2.6.18 but facing problem of
modutils and iptable.
Can anybody help me out of this.
Thanx and Regards
sanchal singh
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Linux your Life, Don't Window it [[]]
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Re: [asterisk-users] inband DTMF for g729
inband is for G711 (uLaw) only. Try rfc2833 Jon Weisman | Sales Engineer International Bell Communications www.ibell.net - Original Message - From: Matthew Fredrickson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 22, 2007 3:20 PM Subject: Re: [asterisk-users] inband DTMF for g729 Sounds like you need a new SIP carrier. G.729 has a way of destroying inband DTMF tones. --- Matthew Fredrickson Software Engineer Digium, Inc. On Jun 22, 2007, at 1:20 PM, Gary Chen wrote: Does anybody know why Asterisk does not support inband DTMF for G.729? Our SIP carrier use inband dtmf for G.729. This causes problem for us to use it for our Asterisk IVR system. Any suggestion to solve this problem? Gary ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX over T1
I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1
On Fri, 2007-06-22 at 17:43 -0400, Joe acquisto wrote: I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? This is what you are looking for: http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX over T1
Thats exactly what i would do. install a channel bank on asterisk with an fxs card in it and using option D of the dial app you could do DID routing On 6/22/07, Joe acquisto [EMAIL PROTECTED] wrote: I have an existing Hylafax system using a mainpine 4 port board, 4 POTS lines. Have a recently installed Asterisk system, with a dedicated T1 line. (Well, it's really a fonality system). What would I need to do, or where is the reading material, for what I need to do, to convert the Hylafax server to use the T1 line? Reliably. Preferably to use DID's as well. The current FAX works fine, but there is some desire to get rid of the analog lines. Could one add some sort of device in the Asterisk server, to act as FAX extensions, keeping the mainpine on the hylafax? Like a TDM400p with FSX modules? I'm just saying, ya know? I suppose I have to ask fonality, since it's their box? joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does Early Media have to be Ulaw?
On 6/22/07, Douglas Garstang [EMAIL PROTECTED] wrote: I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid=Test hone 1 +19256002182 host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default insecure=port dtmfmode=rfc2833 canreinvite=yes qualify=yes disallow=all ;allow=ulaw allow=g729 Level 3 sends early media… --- Transmitting (no NAT) to xxx.yyy.34.210:5061 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.yyy.34.210:5061;branch=z9hG4bK-tenor-d802-22d2-004d;received=xxx.yyy.34.210 From: sip:[EMAIL PROTECTED];tag=d80222d2-27 To: sip:[EMAIL PROTECTED];tag=as4fe079a5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] ontent-Type: application/sdp Content-Length: 261 v=0 o=root 2235 2235 IN IP4 xxx.yyy.34.195 s=session c=IN IP4 xxx.yyy.34.195 t=0 0 m=audio 10484 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv and Asterisk responds on the console with: [Jun 22 10:06:03] WARNING[32573]: channel.c:2882 set_format: Unable to find a codec translation path from g729 to slin [Jun 22 10:06:03] WARNING[32573]: indications.c:121 playtones_alloc: Unable to set 'SIP/19256002182-096ac918' to signed linear format (write) This doesn't happen when progressinband=no. It almost seems like Asterisk has to do early media as G711 only. Is that the case??? Doug. Doug, The SDP says it is G729 (no surprise there). It looks like Asterisk is trying to transcode that to slin from g729. What dialplan logic is this going into? Can you post the section from extensions.conf? My guess is that some application in Asterisk (Dial, Queue, something) is trying to generate ringing (playtones_alloc from indications.c is a dead giveaway) but the call fails because you don't have a g729 codec installed and can't transcode from slin (ringing). Just a guess... -- Kristian Kielhofner ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] install Asterisk-addons 1.4.2
I have Asterisk 1.4.5 and addons 1.4.1. Can anyone tell me if I can just install addons 1.4.2 on this system without re installing Asterisk? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clive.chan(Alpha Trilogies Networks) Sent: Wednesday, June 20, 2007 9:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] install Asterisk-addons 1.4.2 Hi, I am trying to install the Asterisk-addons-1.4.2, and when I make install it prompt me such error messages make[1]: Entering directory `/usr/src/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory make[1]: *** [install] Error 1 make[1]: Leaving directory `/usr/src/asterisk-addons/asterisk-ooh323c' make: *** [install] Error 2 How to solve it out? clive chan Alpha Trilogies Networks Sdn Bhd Tel : 04 - 647 288 Ext: 338 Tel : 04 - 647 2999 Mobile : 012 - 408 6376 email : [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] modules loading
Hi all, Recently I am trying to install the Asterisk 1.4, I has some error while loading the following modules, can some one help on those issues? Error during loading the modules; Basically, chan_ooh323.so, and res_config_mysql.so [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'chan_ooh323.so': /usr/lib/asterisk/modules/chan_ooh323.so: undefined symbol: ast_rtp_bridge [Jun 23 12:10:01] WARNING[30257] loader.c: Error loading module 'res_config_mysql.so': /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: option_verbose If I manually disable the modules in the modules.conf then my Asterisk 1.4.5 will run. I am using belwoing release; Asterisk 1.4.5 Zaptel 1.4.3 Asterisk-addons 1.4.2 Libpri 1.4.0 Thank you in advance if some one can help. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Single ringer phone for incoming calls, that anyone can answer
Hi list, Does anyone have any advice on the following: Incoming calls to our office come in on a SIP trunk. Since all our offices/desks are in close proximity, we would like just a single phone to ring when a call comes in instead of ringing every person's phone. Currently we've got this working by having all the phones in a callgroup/pickupgroup and incoming calls ring the 'ringer phone' extension, then we can use the *8 to pickup the incoming call from any other phone. The problem though, is that if two people in the office call each other, *8 from a third phone also picks up their call, which is not the desired effect. So in essence, I'm asking whether there's a better way to pickup an incoming call from our external SIP trunk, whilst its ringing only a specific extension, without picking up overlapping internal calls? Regards, Tom ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
