Re: [Asterisk-Users] Do people actually answer questions here?
Its not that the answers aren't out there, nobody bothers looking for them. Being a relative new-b on Asterisk, I have to agree most of the information is available on several Internet pages. However, information is scattered around on many pages, and for someone who isn't familiar with stuff like 'codecs' and VoIP in general (and maybe even Linux for a start), it can be hard to find the right information. But instead of complaining about the lack of -findable- documentation, one can try to enhance existing documentation. That's the power of Open Source. As I am not a coder, I'll be trying to help the community by making the documentation better, especially for 'new bees' like me. Bzz, Ralf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Chan_Capi Down
Hi all, are you able to see incoming calls at the isdnlog? I have guessed I have a problem with the capi/isdn/card itsself and not really with asterisk. Felix Thanks I will give that a try. Looks like this may need a bug report? We are all getting the same errors. Outgoing is fine for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: 28 June 2004 23:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: Chan_Capi Down Same here :-( asterisk show's this error in the same moment i'm trying to pick up an incoming call: Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 This problem starts with cvs update -D 6/21/04 21:00:00 CET If i revert back to cvs update -D 6/21/04 18:00:00 CET the problem is gone. -- original message -- I am also having the same problem. Latest CVS Latest Capi When it does work and you pick up the phone, CAPI disconnects the call. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ePyron Felix Deierlein Sent: 28 June 2004 18:34 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Chan_Capi Down Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI -- data = @89930:0107901723168212 -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/2 == CAPI Call CAPI[contr1/89930]/2 -- CONNECT_CONF ID=003 #0x000d LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_CONF ID=003 #0x000f LEN=0014 Controller/PLCI/NCCI= 0x Info= 0x2002 -- DISCONNECT_IND ID=003 #0x0003 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, h, 1) exited non-zero on 'SIP/ePfd-7515' dmesg shows: isdn_dc2minor: di(0) ch(-1072539760) invalid capidrv-1: now up (2 B channels) capidrv-1: D2 trace enabled capi: controller 1 up kcapi: notify up contr 2 capidrv: controller 2 up isdn_dc2minor: di(1) ch(-1072539760) invalid capidrv-2: now up (2 B channels) capidrv-2: D2 trace enabled capi: controller 2 up kcapi: notify up contr 3 capidrv: controller 3 up isdn_dc2minor: di(2) ch(-1072539760) invalid capidrv-3: now up (2 B channels) capidrv-3: D2 trace enabled capi: controller 3 up kcapi: notify up contr 4 capidrv: controller 4 up isdn_dc2minor: di(3) ch(-1072539760) invalid capidrv-4: now up (2 B channels) capidrv-4: D2 trace enabled capi: controller 4 up I hope, that you could help me... Thanks Felix Deierlein _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, recently wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services. You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base. This is how Asterisk grows, free contributions and consultants that are paid to add functionality on a case by case basis. ** Be a community member - contribute! The Asterisk software growth is very much based on user contributions. That's really how we all pay for the software - and get revenue back. If you develop custom functionality, you can rest assured that there is someone out there that wants it, needs it and will be helped by it. Don't forget to contribute. Open Source is both giving and taking. The financial model behind it all is really cooperative in some way. As one member to the community said to a contractor: Hey, I'm paying you to deliver code to me, then I'm giving it away to the community. How did this happen? It's the Open Source business model. And if it didn't work, we wouldn't have a lot of the software platforms that we all use in our business systems - Linux, Apache, MySQL, PostgreSQL and Asterisk. ** Remember: It's Open Source, it's voluntary Asterisk.org is a Open Source project. This means you can't request help from people, demand new functions or support. However, there are many individuals and companies
Re: [Asterisk-Users] Re: 'a' and 'o' extensions do not work with app_voicemail.c (was: Newbie needs help)
As I said, I'm not an expert, so I would strongly recommend against committing this as-is... someone please interpret why this works and fix the root problem (or help me understand why this works so I can fix the root problem). I'd suggest that you open a bug with you problem and your patch to bugs.digium.org. Please be sure to make it with cvs diff -u or, better, add the line diff -u to your ~/.cvsrc file. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * Busy-Redial ??
I was wondering if anyone knew of a way to create a busy-redial feature in the * dialplan? For example, you try to call 12125551212 but the number is busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two channels (maybe even with a distinctive ring). If anyone has any suggestions on how to accomplish this please let me know. Thanks in advance, William Mandra Chief Technology Officer M-networks.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cannot make app_prepaid
try to vi the app_prepaid.c and there is a line #include postgresql/libpq.h and edit it and make #include libpq.h this problem exist in modifyed app_prepaid only. Best Regards Hekuran Doli Eureka... i must edit the Makefile hehe... yes correct, i must mantion about the psql lib -L /usr/local/pgsql/lib Thnx. Regards, Freddy Setiawan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Joshua McClintock Sent: Tuesday, June 29, 2004 12:56 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] cannot make app_prepaid I believe libpq is a postgres dev library. You probally need a psql-dev package of some sort. - Original Message - From: Freddy Setiawan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 28, 2004 9:34 PM Subject: [Asterisk-Users] cannot make app_prepaid hai there, today i tried to implement the prepaid application to my * box. I do the step that mantion in to voip-info. i copy the app_prepaid.c and Make file to my asteris/apps, then i run the make. but it show an error like : gcc -shared -Xlinker -x -o app_prepaid.so app_prepaid.o -lpq /usr/bin/ld: cannot find -lpq collect2: ld returned 1 exit status make[1]: *** [app_prepaid.so] Error 1 make[1]: Leaving directory '/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Any sugestion what should i do? Best Regards, Freddy Setiawan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Jeremy McNamara wrote: Michael Manousos wrote: The performance of the oh323 channel driver is limited by OpenH323. asterisk-oh323 uses the (more complete) RTP implementation offered by the library, and not that of Asterisk. Of course there are pros (adaptive jitter buffer, RTCP implementation) and cons (lower performance). It's up to the user to select the one that performs better for his application. flamePut the crack pipe down./flame I won't bite. We all know what you have done. We have gone over this before, asterisk-oh323 is limited by the method you implemented to buffer the audio around. Jeremy McNamara Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Security Vulnerability in Asterisk
Jim Rosenberg wrote: --On Monday, June 28, 2004 7:21 PM +0200 Michael Sandee [EMAIL PROTECTED] wrote: Other than that... if these problems are not being published when fixed... then other distro's do not have a chance to fix it... (think about distro's that use stable code, but haven't updated to 0.9 because of problems) I have to say -- with somewhat less vehemence -- that I'm another user who sure never noticed that the stable release of Asterisk had moved from 0.7.2 to 0.9x. This should have been an important announcement on *SEVERAL* security grounds. As of 0.7.2, the recommend version of channel H323 had some very serious vulnerabilities that the OpenH323 folks had fixed months previously. The latest versions of asterisk-oh323 use OpenH323 1.13.5, Pwlib 1.6.6. Why don't you use that one? This is an opportune time to repeat: H.323 uses ASN.1. ASN.1 is fiendishly complex and is a known bad boy in which many security holes have appeared over the years. It would be naive to think there won't be more. As VOIP hits the big-time and Asterisk joins the ranks of some of the other more famous open-source projects, quick response to security vulnerabilities will be expected. It's nice to know in the case of these format string problems that they were in some sense addressed promptly, but we're not all subscribed to the dev list. A vulnerability that is fixed in CVS head but not back-patched to stable *is not fixed* as far as a large percentage of the user base is concerned. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323, new version 0.6.3
Florin Andrei wrote: On Mon, 2004-06-28 at 07:45, Michael Manousos wrote: Hello all, Bugfix release 0.6.3 is now available. Basically, call indications should work ok now. Also, the OH323 channel variables for incoming calls are set properly (they can be used for special authentication purposes). Download: http://www.inaccessnetworks.com/projects/asterisk-oh323 Will it work as a H323 gatekeeper? No, if you want gatekeeper functionality from your Asterisk box, just run gnugk. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * Busy-Redial ??
Hi, -Original Message- I was wondering if anyone knew of a way to create a busy-redial feature in the * dialplan? For example, you try to call 12125551212 but the number is busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two channels (maybe even with a distinctive ring). If anyone has any suggestions on how to accomplish this please let me know. I have not implemented this, but it should be doable by creating a file in the spool directory. You really just need to tinker around with contexts and actions to do 'the right thing' (tm). An AGI script to generate the spool file would probably do the trick. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] P32mxi
Can anyone confirm if they have this channel bank running on Asterisk? There is a post in the archive about having a few niggley problems but no follow up. I tried to email the guy but the mail address is bouncing. I'm not having much luck finding any of the archive recommended channel banks cheap in the UK. I really want to try and get someone that someone has already had success with (being a numpty). I am finding no reference to the 650 /750's that everyone is suggesting. Ill keep looking, cheers Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Sipura SPA-1000 configs
Anyone had any experience here on how to config both ends, asterisk and the sipura SPA1000 TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] linux kernel 2.6.6
Hi All trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 as going from the readme. is 2.6 not compatiable with asterisk and should I go back to 2.4.26. Also has anyone got the sipura 3000 working with asterisk, both fxo and the fxs ports on the unit. regards Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Do people actually answer questions here?
Jean-Yves Avenard [EMAIL PROTECTED] wrote: I've only been watching this list for the past 2 days. And it seems to be an one way street: -Tell about your problems and what you would like to do. Usually no answer. I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Personally, I only read about 20% of the articles in this list. I delete whole threads based solely upon the Subject line, especially when it's set to (no subject) or has no clear indication of content. If I've read a couple of articles in a thread and have decided that I'm not interested, I tend to blindly delete all followups. This means that if someone asks a new question by following up to an existing thread then it'll probably get caught up in my mass delete and won't be seen by me at all. I said that I tend to do this, I don't always do it - obviously. :-) I also have very little interest in top-posted followups and HTML emails, and often won't bother read past the first sentence. This is a response to the original posters' laziness rather than anything else, although a couple of them will be of enough interest for me to ignore these annoying breaches of netiquette and read the entire article. This is just my personal policy on the matter, but I suspect that others do the same. When you're subscribed to several high volume mailing lists, there's not enough time to read everything. Having said all of that, two days is not really enough time to monitor a mail list before giving up on it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] linux kernel 2.6.6
On Tue, 2004-06-29 at 17:43 +1000, yaboo wrote: Hi All trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 as going from the readme. So ln -s /usr/src/your source directory /usr/src/linux-2.6 is 2.6 not compatiable with asterisk No (2 negatives = positive) Yes * and 2.6 do work perfectly and should I go back to 2.4.26. Only if you can't follow the above. :) -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] linux kernel 2.6.6
yaboo [EMAIL PROTECTED] wrote: trying to compile asterisk under linux kernel 2.6.6. Currently under zaptel get the following error make linux26 Link /usr/src/linux-2.6 to your kernel sources first! make: *** [linux26] Error 1 Type this: # cd /usr/src # ln -s linux linux-2.6 That'll link linux-2.6 to your kernel sources. Error messages are your friend. is 2.6 not compatiable with asterisk and should I go back to 2.4.26. I'm using 2.6.7-gentoo-r6 and it works very well. I assume you're using the latest Zaptel and Asterisk from CVS. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap X100P oscillation
Thanks for the responses. I have tried it with aggressive cancellation both on and off. I think that on helps a tiny bit. I'm glad that Mike Benoit as seen something similar, but of course sorry that he is suffering like me! It is worse when I have a Phone-switch-X100P-IAX-Internet-IAX-X100P-switch-Phone link set up. The other thing which may have helped a bit is using a large set of IAX Jitter buffers. It may be the latency which is helping though, rather than the anti-jitter aspects. Peter -Original Message- From: Brian McSpadden [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 22:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap X100P oscillation Try recompiling your zaptel package without the aggressive echo cancellation enabled. I have aggressive cancellation help before, I but I have also seen it hurt things before. Brian On Mon, 28 Jun 2004 16:26:32 +0100, Whisker, Peter [EMAIL PROTECTED] wrote: I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip to isdn-capi call problem
Hi Tomaz, make sure you disable the G723.1 codec in your SIP device, asterisk does not support G723.1. Use G711 (alaw, ulaw)! best regards Klaus -- Klaus-Peter Junghanns CEO, CTO Junghanns.NET GmbH Breite Strasse 13a - 12167 Berlin - Germany fon: (de) +49 30 79705390 fon: (uk) +44 870 1244692 fax: (de) +49 30 79705391 iaxtel: 1-700-157-8753 http://www.Junghanns.NET/asterisk/ Am Mo, 2004-06-28 um 10.52 schrieb Tomaz: anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI -- Executing Dial(SIP/102-767c, CAPI/2:5) in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making progress passing it to SIP/102-767c Jun 28 10:51:21 NOTICE[278545]: channel.c:1654 ast_set_read_format: Unable to find a path from G723 to ALAW Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from ULAW to G723 -- CAPI[contr1/2003002]/0 is ringing Jun 28 10:51:21 WARNING[278545]: chan_sip.c:1788 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 8/4) Jun 28 10:51:21 WARNING[278545]: channel.c:1485 ast_prod: Prodding channel 'SIP/102-767c' failed Jun 28 10:51:21 NOTICE[278545]: channel.c:1621 ast_set_write_format: Unable to find a path from SLINR to G723 Jun 28 10:51:21 WARNING[278545]: indications.c:76 playtones_alloc: Unable to set 'SIP/102-767c' to signed linear format (write) -- CAPI Hangingup == Spawn extension (from-sip, 9, 1) exited non-zero on 'SIP/102-767c' http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap X100P oscillation
I find that if I drop the RX gain too much I start to lose DTMF decoding. The Asterisk calls lose at least 3-6db end-to-end compared with a normal call. If I bring the gain up, the symptoms sound exactly like yours. The gain I am using is more like Rx=-2, Tx=0 but this is still quite quiet. I guess that line impedance mismatch between US and European standards accounts for some of the gain loss. Peter -Original Message- From: Mike Benoit [mailto:[EMAIL PROTECTED] Sent: 28 June 2004 21:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Zap X100P oscillation I wonder if your issue and mine are related somehow. I have a asterisk server with 4 FXO cards in it, and when a call comes in one ZAP channel, then dials out another, I hear what could be described as a steam engine starting up. It starts off kinda slower/ quiet, then quickly (in about 2-4 seconds) completely over powers the line. The only way I could stop it was by adjusting the gains. rxgain=-8.5 txgain=4 Seemed to do the trick. As did: rxgain=-6.5 txgain=1 An rxgain of even -8.0 or -6.0 in either case would result in this steam engine sound. -8.5 or -6.5 would make it go away completely. I'm using a CVS checkout from yesterday, and I tried with both echotraining=800 and turning echo cancellation off completely. Neither made any difference. It would be really nice to be able to use a positive rxgain value. I haven't tried with the echo app, but using just one FXO card works fine with almost any rx/txgain value. As soon as the call utilizes two FXO card at the same time, the steam engine sound occurs. On Mon, 2004-06-28 at 16:26 +0100, Whisker, Peter wrote: Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles down but it is very very quiet. I have tried the latest CVS Head with echotraining=800 set and also complied with the aggressive echo cancelling, but nothing seems to help. Ideas welcome! Many thanks Peter Whisker This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cvs log archive
Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. Thanks Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Chan_Capi Down
Hiya, Looks like this may need a bug report? We are all getting the same errors. Sure, but i guess a bug to bugs.digium.com will be rejected, chan_capi is not in CVS. Maybe [EMAIL PROTECTED] could do a fix, :-D PLEEASE :-D? BTW: kapejod, any chances to disclaim chan_capi to digium? It would safe some troubles if it was in CVS... Outgoing is fine for me. yes, no problem with outgoing. bye. aa _ Need more speed? Get Xtra JetStream @ http://xtra.co.nz/jetstream ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Customized Call Parking
Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1 user 2 VM boxes?
Hi, Would be interestd in anyones ideas for this problem.. We are starting a new division to our company, the people in this new division will be the same people who are on the old division.. Calls for each division come in on seperate numbers and go through seperate menus but ring to common extensions, this is easy enough.. The problem is with Voicemail.. I need to have seperate VM boxes for each division so that the user will be able to distinguish between the messages for each division.. Basically the only ways I can think of to make this work is to teach the users how to access each seperate VM box with a seperate VM box number and password or I could use VM contexts and have two identical VM boxes in each context but then the user will have to access VoiceMailMain differently (eg 100 for division1 VM and 101 for division2 VM) for each VM box.. Can anyone think of any easier ways? later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 stops to send/receive calls
Jorge Mendoza wrote: Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? We had one do the same thing. Changing the registration timeout in the phone.cfg file down to 20 seems to have fixed the problem. -rb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1 user 2 VM boxes?
Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs log archive
Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. You can use GUI tools like cervisia. Or use cvs2log script or so. Just do your usual google searching or look for third-party apps on CVS's home page. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1 user 2 VM boxes?
Senad Jordanovic wrote: Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ The single server works fine for the two divisions making and recieving calls.. Its that each extension needs two VM boxes (one for each division) thats the problem and I want to make it as simple as possible for the users to access the two seperate VM boxes from the one extension.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ruggedised IP Phone
Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. Best regards Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cvs log archive
Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. You can use GUI tools like cervisia. Or use cvs2log script or so. Just do your usual google searching or look for third-party apps on CVS's home page. You'll find that the log comments are not a lot of use. You generally have to dive into the actual change to find out what oops means this time. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1 user 2 VM boxes?
At 11:40 29/06/2004 +0100, you wrote: Senad Jordanovic wrote: Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ The single server works fine for the two divisions making and recieving calls.. Its that each extension needs two VM boxes (one for each division) thats the problem and I want to make it as simple as possible for the users to access the two seperate VM boxes from the one extension.. Why not front the VM access number with an IVR press 1 for Division 1 press 2 for division 2 Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voip Account over H323
I have a VOIP account including a telephone number in another country. The connection uses the H323 connection to connect to the remote PBX. Can I learn Asterisk to use that connection for outgoing calls and also that he can handle my incoming calls? Johannes
Re: [Asterisk-Users] 1 user 2 VM boxes?
Jason Williams wrote: At 11:40 29/06/2004 +0100, you wrote: Senad Jordanovic wrote: Can anyone think of any easier ways? How about if you put second division on different server, and then share VM storage on the network between two asterisk boxes? SJ The single server works fine for the two divisions making and recieving calls.. Its that each extension needs two VM boxes (one for each division) thats the problem and I want to make it as simple as possible for the users to access the two seperate VM boxes from the one extension.. Why not front the VM access number with an IVR press 1 for Division 1 press 2 for division 2 Thats not a bad idea.. Then have two identical VM boxes in seperate VM contexts with the same password.. That should work.. Now I just have to customise the email thet is sent to tell the user which divisions mailbox is the one with the message but that should be easy.. Thanks to all.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling libiax2 on windows
Hi! I'm trying to compile libiax2 on windows using msvc6. In the libiax2\src\iax.c file, line 670, I'm getting a: error C2229: struct __unnamed has an illegal zero-sized array It seems to complain due to the last member of iax_frame. Does anybody knows what should I do to make it compile? FWIW, it compiles fine using gcc on cygwin. Thank you in advance for any tips! Cheers, = Joaquin Cuenca Abela e98cuenc at yahoo dot com __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ruggedised IP Phone
Matt, After much searching, I could not find any Ruggedised IP Phone's out of the box... I am now looking at using a Stunning Art Deco (:-) Dorfone (GBP300) from www.spicecommunications.co.uk/shop/Door_Entry_Systems.htm This range of speaker entry-phones, sit between your BT line and handsets; when the buzzer is pushed the handsets ring. I am guessing that this can be connected to a FXO module on my TDM400P. My other option, was to cannibalise a broken Cisco 7940, putting the innards into a cheap entry phone e.g. stock number 227-7892 from www.rswww.com for GBP50 Connecting up the mic, speaker and one of the speed-dial buttons! Though, I guess, if this worked, it might make a neat niche product Rgds, Robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Sent: 29 June 2004 11:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ruggedised IP Phone Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. Best regards Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ruggedised IP Phone
I am now looking at using a Stunning Art Deco (:-) Dorfone (GBP300) from www.spicecommunications.co.uk/shop/Door_Entry_Systems.htm This range of speaker entry-phones, sit between your BT line and handsets; when the buzzer is pushed the handsets ring. I am guessing that this can be connected to a FXO module on my TDM400P. Cheers Robert, I'll give them a call and see just how the phones work. I was interested in the SlimLine Version beneath the Dorphone on the URL you mention. I'm looking for a solution for a large house, the gate is about 25 meters from the main house. I was hoping to set the system up so that an extension could be dialled when someone picks up the phone; that extension rings in the main house and also in surrounding outbuildings. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matt Sent: 29 June 2004 11:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Ruggedised IP Phone Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. Best regards Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users PLEASE READ: The information contained in this email is confidential and intended for the named recipient(s) only. If you are not an intended recipient of this email you must not copy, distribute or take any further action in reliance on it and you should delete it and notify the sender immediately. Email is not a secure method of communication and Nomura International plc cannot accept responsibility for the accuracy or completeness of this message or any attachment(s). Please examine this email for virus infection, for which Nomura International plc accepts no responsibility. If verification of this email is sought then please request a hard copy. Unless otherwise stated any views or opinions presented are solely those of the author and do not represent those of Nomura International plc. This email is intended for informational purposes only and is not a solicitation or offer to buy or sell securities or related financial instruments. Nomura International plc is regulated by the Financial Services Authority and is a member of the London Stock Exchange. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing incoming H.323 calls to specific contexts.
Hi, We've been working a lot with Asterisk in SIP for over 6 months but I've finally succumb to the pressure of H.323. I need to find a way to do what we do with SIP but with H.323. That is to have calls from H.323 peers placed into their own unique context (unique to the endpoint placing the call into Asterisk) within Asterisk so this is obviously done using REGISTER's within SIP but trying to do this with H.323 seems more challenging. I've installed GNUGK and have successfully had a H.323 device authenticate with the GNUGK and place calls onwards to Asterisk but I am unable to figure out how to place those calls into a unique context per H.323 endpoint/device/account without using their CLI to do so. I'm sure the community have solved this issue before, any help would be much appreciated and example configs would be perfect. Thank you in advance, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h323 audio problem (next)
Hello everybody, we updated yesterday the full cvs version which include the h323 modification NoFastStart = TRUE in ast_h323.cpp So call to our GK EP are again working. But we also connect to a gw which need FastStart. So there, calls are still without audio. Thanks for any hint -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do people actually answer questions here?
On Tuesday 29 June 2004 02:42, Ralf Van Dooren wrote: But instead of complaining about the lack of -findable- documentation, one can try to enhance existing documentation. That's the power of Open Source. As I am not a coder, I'll be trying to help the community by making the documentation better, especially for 'new bees' like me. This is where I have a problem. The documentation is centered on voip-info.org and on Asterisk's plainly marked Documentation link. It's been 32000 messages since I've signed up but I am *positive* that BOTH links are provided on the autoresponder when you sign up to this list. How crystal-effing-clear must things be for people to go and look for themselves before complaining on this list? Must we make the list moderated and autorespond with pre-fab Google searches that the asker simply has to click on and save themselves the trouble of writing the query into Google's search window themselves? I'm serious here -- voip-info.org's search engine works. Google works. For newbies yes they may have some trouble with the incantations but that's why they should not be diving in headfirst and then bitterly complaining that nothing works on the list. READ, dig around voip-info and asterisk's site. There's a WEALTH of knowlege there and most of it is not cryptic. A lot of it is even geared to newbies. Perhaps a glossary would be helpful but every day I start to think that basic research skills should be a prerequisite before being allowed to play with any OSS project. It's frustrating for the newbie, frustrating for the experts and all around a bad thing. You'll never have enough documentation to satisfy some people because they don't want to educate themselves; they want to ask questions and get personalized answers. We do that too, for a price. That's what consulting is all about. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Sipura SPA-1000 configs
tucker scribbled on Tuesday, June 29, 2004 2:37 AM: Anyone had any experience here on how to config both ends, asterisk and the sipura SPA1000 There is accurate information out there if you do a google search. The first hit or two will have what you need. I don't remember if the example I used as a base is on the wiki or another site. I can try to remember to look at my bookmark list when I get home if you do a search and can't find what you need. All in all, it is pretty straight forward, I only used the example to to see if there were any oddball settings I needed. Just set up your peer/user/friend (whichever you are doing) in your sip.conf as you would with any other SIP device. Punch in your Asterisk connection and authentication settings on the SPA's management page, and you should be set. Let me know if you can't get it working, and I can send you the relevant portions of my sip.conf and screenshots of the admin page on the device itself. If you need this, send an e-mail to my home address hall.jeremy at gmail dot com. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call dropping out after 5s: Solution!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm the one who posted a message about the fact that nobody answer anymore to questions asked. I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config Well, it wasn't (as I was expecting). I compiled Asterisk under a Linux RedHat 9 PC, I copied across all the config files from the FreeBSD server to the Linux PC. Started asterisk, registered the phone: all worked fine first go, and guess what?? No drop after 5s. I guess the FreeBSD support is just not up to scratch. So if anybody is having this problem, just try to find an old PC somewhere load linux and up you go. This will raise the cost of our Asterisk installation as we now have to include an extra PC (all our servers are FreeBSD). Many people have responded to my complaint that there was a lot of information available and basically people should just read it. My point is: do not always assume people did not read available documentation. For the past 4 days I've read the almost entire wiki site ; I looked at almost all messages sent to this distribution list that was somehow related to my problem: no luck. I also faced another problem today. Encouraged by the progress with the Linux server, I configured the TDM03B (3 FXO ports). Loaded the kernel module, no problem card was recognized. Started asterisk, tried to make a call: - - -Can start Zap channel try about a dozen different configuration ; no luck. As I ran into similar problem with my FreeBSD box, I thought: there's no way it can be the driver this time (TDM driver on FreeBSD is an early beta) So I opened the Linux PC, moved the card from one PCI slot to another. Tried again as above: no luck. Open the PC again, move the card into the third and last PCI slot. Disabled the USB 2.0 host controller that was sharing the same IRQ (7). Start asterisk, dial: It works! So now which PCI slot your card is in makes a difference! Something very fishy here. Was this documented? I didn't find anything like that whatsoever. I guess this could be added somewhere Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA4XByXeDVKqIr3GURAj5tAJ9mpDhS5kq/us6IY4nAUAGxe2yBKgCggr+u z9j5LzrCucUO9obZsrTQ/Eo= =F9Ge -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to test E1 interfacing?
Hi Tony- Two E1's running on a TE405P should be a better choice, as the TE405 is capable of bus mastering which gives better performance. Besides, you only use one PCI slot in this configuration. However, of course you lose board redundancy. You can run 1 E1 port to another by using a crossover cable, as follows: 1 -- 4 2 -- 5 4 -- 1 5 -- 2 You can then use the outbound call generation capability of asterisk (see the file called sample.call) to generate calls on 1 span, and receive them on the other with the normal dialplan. I've done this many times for load testing. You can do it on one machine no problem. There's a little more information about this on the Wiki: http://www.voip-info.org/wiki-Asterisk+dimensioning Regards, Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Tuesday, June 29, 2004 2:00 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to test E1 interfacing? Hi, I have a project coming up which will need to interface Asterisk to E1 trunks in the UK. I have a couple of questions which I hope someone can answer, or give me some pointers: 1. If I want two E1 trunks, is there anything to choose, performance-wise, between using two ports on a single TE405P, and using two E100P cards? 2. How can I test the E1 operation in the lab, which doesn't have an E1 line available, before taking the unit to the installation site? Can I run two Asterisks back-to-back? Can I run one port into another on a single TE405P? I couldn't find anything on the above in the Wiki; if I didn't look hard enough, please tell me where I missed. Thanks. Any advice would be gratefully received! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs log archive
On Tue, 2004-06-29 at 04:43, Chris Stenton wrote: Ok I'm no cvs expert is there a cvs command to get a date sequential cvs log archive for cvs-head or a URL for it. With so many daily changes its hard to keep track of what the changes are. Go to http://lists.digium.com/ pick the link that says Asterisk-Cvs CVS Updates to Asterisk and the Core Components. Subscribe or just browse the archives. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] General advice on confs and setup for new users
tucker scribbled on Sunday, June 27, 2004 8:56 AM: This is what I want to do, however I am not sure how to achieve it, can you help? Asterisk is running with X100P card to local PSTN Allow incoming calls over the internet to Asterisk Allow internet calls to dial out (restricted) using X100P Allow incoming calls via X100P to dial extension over the internet Allow incoming calls via X100P to be directed to internet extension N after set time and/or on no reply This is actually a pretty simple configuration. Do you already have your X100P card installed and set up in your system? If you do, use the 'make samples' or similar command that Asterisk points you to when you compile. Go through the config files you will need for your setup. zaptel.conf for your X100P, sip.conf and/or iax.conf for your remote extensions, and extensions.conf for your dialplan. Make sure you understand what the samples are doing, and look for other examples to confirm your understanding. Then start modifying it to meet your needs. There is so much to read, not sure where best to start That is a very common complaint from people starting out. Like I said above, just start out simple. Use the wiki and google, and take it slow. Make it so when a call comes in on the PSTN, it calls one local extension. Then add another. Then set it up to allow one remote extension to dial out local calls only, and another to do long distance. Then set up a basic IVR (voice menu) asking the caller to press 1 for your first phone, or 2 for your second one. One of the biggest complaints I have about newbies, is their expectations of installing it, and having it work out of the box for their specific hardware and situation with little to no learning and configuration. I've seen people here and in the IRC channel that complained when they practically want a multiple server clustered call center system to work with their BrandX telephony hardware, when they have never used Asterisk at all. Most of the time the hardware they want to use is not supported, and there is plenty of documentation on the archive stating as such. Like I said, you have a very simple system to start out with, which is good. If you run into snags, feel free to ask me on IRC (jjhall is my nick) or just ask the channel in general. I'm on about half the evenings out of the week, and usually try to at least monitor here and there throughout the day while at work. Just /msg me if I am on and I will answer when I get a chance. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ruggedised IP Phone
On Tue, 2004-06-29 at 05:42, Matt wrote: I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. Any comments on your experiences would be very much appriciated. http://www.google.com/search?hl=enie=UTF-8q=site%3Alists.digium.com+doorphonebtnG=Google+Search -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call dropping out after 5s: Solution!
On Tue, 2004-06-29 at 08:36, Jean-Yves Avenard wrote: I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config [snip] So now which PCI slot your card is in makes a difference! Something very fishy here. Was this documented? I didn't find anything like that whatsoever. I guess this could be added somewhere cat /proc/interrupts will tell you what IRQ your card is on and if any other devices are on the same IRQ. Usually calls dropping is caused by callprogress=yes or busydetect=yes (the beaten, bloody, dead horse is in the archives). If your card was sharing IRQs then the expected symptom would be poor audio quality, not dropped calls. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing CallerID on PRI problems
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600 regardless of the extension that
makes the outgoing call.
We have tried using the variable SetCallerID(${BYEXTENSION}) but still
get the same results.
Any suggestions?
JP McInnis, Director of Technology
Copiah Lincoln Community College
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Re: [Asterisk-Users] Call dropping out after 5s: Solution!
Jean-Yves Avenard wrote: I posted two days ago a problem I was facing that calls made over the Internet to my Asterisk gateway would hang-up after just 5s (no NAT were involved) Answer I got was: it's your config Well, it wasn't (as I was expecting). I compiled Asterisk under a Linux RedHat 9 PC, I copied across all the config files from the FreeBSD server to the Linux PC. Started asterisk, registered the phone: all worked fine first go, and guess what?? No drop after 5s. I guess the FreeBSD support is just not up to scratch. I would consider the operating system under which you are trying to run part of the overall config. On that note: which part of Asterisk - The Open Source LINUX PBX do you not understand? Many people have responded to my complaint that there was a lot of information available and basically people should just read it. My point is: do not always assume people did not read available documentation. For the past 4 days I've read the almost entire wiki site ; I looked at almost all messages sent to this distribution list that was somehow related to my problem: no luck. Problems running Asterisk on [non-Linux] are regularly documented. I guess your search-foo isn't as strong as your bitch/moan-foo. I also faced another problem today. Encouraged by the progress with the Linux server, I configured the TDM03B (3 FXO ports). Loaded the kernel module, no problem card was recognized. Started asterisk, tried to make a call: snip Open the PC again, move the card into the third and last PCI slot. Disabled the USB 2.0 host controller that was sharing the same IRQ (7). Start asterisk, dial: It works! So now which PCI slot your card is in makes a difference! Something very fishy here. Was this documented? I didn't find anything like that whatsoever. I guess this could be added somewhere This comes up on the list at LEAST once per month. Please refer to search-foo vs bitch/moan-foo above. Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net ^ Just not READING it obviously. -- John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Get back a failed transfered call
Hi Folks, I have the following situation: I received an inbound call in my extension A and transferred it to the extension B. But B was busy and I want to capture the call back to my extension. How should I proceed? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing CallerID on PRI problems
On Tuesday 29 June 2004 10:13, McInnis, JP wrote: The monitor shows that the CallerID is being set to the specified number, but yet when the call is received on the user end the ID is always the base number of our DID. For example we have 8600-8650 as DID's but the callerid is always 8600 regardless of the extension that makes the outgoing call. Your telco is restricting your outgoing caller ID. I'd contact your sales rep, as this is something only the telco can fix. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing CallerID on PRI problems
Regardless of what you send in callerid, your PRI has a phone number
associated with it that you don't see, but is used for billing. This is
so you cannot spoof the LD company into thinking the call came from
somewhere other than from you. I believe the PRI provider can provision
the PRI to use either this hard-wired callerid , or the one you provide.
It sounds to me like your PRI is provisioned as the former. I would talk
to your PRI provider and see if they agree and are willing to change this.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 29 Jun 2004, McInnis, JP wrote:
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600 regardless of the extension that
makes the outgoing call.
We have tried using the variable SetCallerID(${BYEXTENSION}) but still
get the same results.
Any suggestions?
JP McInnis, Director of Technology
Copiah Lincoln Community College
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Re: [Asterisk-Users] Outgoing CallerID on PRI problems
Drop the leading 1
On Tue, 2004-06-29 at 09:13, McInnis, JP wrote:
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600 regardless of the extension that
makes the outgoing call.
We have tried using the variable SetCallerID(${BYEXTENSION}) but still
get the same results.
Any suggestions?
JP McInnis, Director of Technology
Copiah Lincoln Community College
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--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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[Asterisk-Users] Re: SIP Softphone
Hi, On Mon, Jun 28, 2004 at 11:00:43PM +0200, Arve Rasmussen wrote: What is the best SIP softphone to use with Asterisk? Really don't know what is the best SIP softphone but I am using linphone with alaw codec and dtmfmode rfc2833. Did not try low bandwidth codecs until now. http://www.linphone.org/ http://simon.morlat.free.fr/download/0.12.2/source/linphone-0.12.2.tar.gz -- Stefan Tichy [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Do people actually answer questions here?
we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outgoing CallerID on PRI problems
Some telcos require you only to send a certain number of digits. Try
sending fewer and fewer digits and see if it starts working. E.g.
instead of sending 1601abcdefg try sending 601abcdefg or abcdefg or defg
or even fg
If this doesn't work, from the CLI type pri debug span x and see what
you get in the SETUP message.
Other possiblities are in zapata.conf - you may need to set the correct
parameters for your type of PRI. Best thing might be to ask your telco
how your line is configured.
Please report back with success/failure. We all like to hear of
people's successes as well as problems!
Rgds
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of McInnis, JP
Sent: 29 June 2004 15:13
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Outgoing CallerID on PRI problems
For outgoing calls made on our PRI circuit we are setting the Caller ID
using the format
Exten = _9XXX,1,SetCallerID(1601XXX)
The monitor shows that the CallerID is being set to the specified
number, but yet when the call is received on the user end the ID is
always the base number of our DID. For example we have 8600-8650 as
DID's but the callerid is always 8600 regardless of the extension that
makes the outgoing call.
We have tried using the variable SetCallerID(${BYEXTENSION}) but still
get the same results.
Any suggestions?
JP McInnis, Director of Technology
Copiah Lincoln Community College
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[Asterisk-Users] Playing the invalid extension input
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
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[Asterisk-Users] Sip Debugging
Hello, When I enable SIP debugging I receive: Peer RTP is at port 10.10.60.16:0 Shouldn't the RTP port be a number between 1 - 2? - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Do people actually answer questions here?
DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! Got that... DO NOT USE GROUP REPLY!!! I will get a copy from the list server just fine without a personal copy to me directly. On Tue, 2004-06-29 at 09:39, Randy Bush wrote: we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. Booo whh, use better tools. Find out how to use Mozilla and tabbed browsing. Tabbed browsing can be set up to make a new tab for any link that goes to a different site. It can also be configured to create new tabs for those annoying popups. Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. If you knew how much time and how much money both of my personal fincances and my employers was put into the knowledge I have currently, you would start to understand why I am so annoyed that you don't seem to want to spend the time it takes to learn. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do people actually answer questions here?
Hi Andrew, I sympathise with your opinion. However if someone was to analyse the messaged in the list they would find that the most basic of questions get most replies. I mean those questions that would take a few minutes to answer searching through the wiki or google. Where questions that are not so straighr forwared get ignored. In my own experiance, every question that I have posted (after hours if not days of searching) has gone ignored. I must add, at no stage though have I felt a reason to complain, as even without answering any of my questions, this list has given me a wealth of knowledge. Thanks Umar. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 29 June 2004 02:42, Ralf Van Dooren wrote: But instead of complaining about the lack of -findable- documentation, one can try to enhance existing documentation. That's the power of Open Source. As I am not a coder, I'll be trying to help the community by making the documentation better, especially for 'new bees' like me. This is where I have a problem. The documentation is centered on voip-info.org and on Asterisk's plainly marked Documentation link. It's been 32000 messages since I've signed up but I am *positive* that BOTH links are provided on the autoresponder when you sign up to this list. How crystal-effing-clear must things be for people to go and look for themselves before complaining on this list? Must we make the list moderated and autorespond with pre-fab Google searches that the asker simply has to click on and save themselves the trouble of writing the query into Google's search window themselves? I'm serious here -- voip-info.org's search engine works. Google works. For newbies yes they may have some trouble with the incantations but that's why they should not be diving in headfirst and then bitterly complaining that nothing works on the list. READ, dig around voip-info and asterisk's site. There's a WEALTH of knowlege there and most of it is not cryptic. A lot of it is even geared to newbies. Perhaps a glossary would be helpful but every day I start to think that basic research skills should be a prerequisite before being allowed to play with any OSS project. It's frustrating for the newbie, frustrating for the experts and all around a bad thing. You'll never have enough documentation to satisfy some people because they don't want to educate themselves; they want to ask questions and get personalized answers. We do that too, for a price. That's what consulting is all about. -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself http://uk.messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Do people actually answer questions here?
Then it sounds like the Wiki is being mis/under-used. The great thing about a Wiki is that if someone has an authoritative answer/solution to a problem, they can just post it up. Personally, I have found the Wiki to have plenty of relevant content within its bounds; more esoteric topircs or one-off issues probably won't be found there easily, especially if someone doesn't think to post up an article on their successes. Greg On Tue, 2004-06-29 at 07:39 -0700, Randy Bush wrote: we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modems behind Asterisk - how?
Hi, Looks like your message got lost in the thread. On 29/06/2004, at 8:47 AM, John Vogel wrote: 1. Use 4 Sipuras (approx. $400). Only problem is, I can't get this to work! Sipura says use the G711 codec but it's not working for me. Anybody have this working? I haven't got any Sipura's to test, but I tried and failed with Grandstream 286's. My general experience (and advice I have been given) is that transferring modem data across a network is a bad thing - any jumps in the call, and any lag at all will cause the whole thing to fail. It's generally OK for fax, but remember that fax works at a significantly lower speed to modem transfer. 2. Use 8 FXS ports (approx. $700). Haven't tried this yet but it is more expensive. I have managed to get a single FXO to FXS working for a dialup modem, although I was only getting 24000bps. I am told (although I haven't had the opportunity to try yet) that when I connect to a digital ISDN line for the incoming calls, I will get better throughput (closer to 56k). That said, this is probably your best option. Andrew smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Re: Do people actually answer questions here?
DO NOT USE GROUP REPLY!!! fat effing chance. fix your mail system. see a very large number of threads. but as you seem unable to look up archives:-), try this in your .procmailrc # prevent dupes # :0 Wh: msgid.lock | formail -D 65536 msgid.cache Booo whh, use better tools. Find out how to use Mozilla and tabbed browsing. been doing that for some years. it can not make up for a bad archive or weak search tools. Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. this is the cult of this is a very complex area. you need to pay us gurus to do it for you. in my 40 years of computing i can not count the fields where i have seen the guru-friendly products and technologies go one of two ways, marginalization and failure or takeover by the massive companies who then marginalize the engineers. i suggest you have another career path planned. If you knew how much time and how much money both of my personal fincances and my employers was put into the knowledge I have currently, you would start to understand why I am so annoyed that you don't seem to want to spend the time it takes to learn. nice of you to turn my comment on the difficulty of the tools into an attack on my knowledge. particularly amusing if you knew more. there are reasons there are so many repeated questions on this list. some of the reasons are weakness in the documentation. we can pretend this is not a problem and attack the questioners or those who try to discuss the weaknesses, or we can discuss how we might address them. clearly you have made your choice. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blank faxes with RxFAX
Patrick J. Conroy wrote: Hello All, I have finally pulled CVS HEAD and built it with app_rxfax and app_txfax to try to solve the problem that I was having with blank faxes. Fortunately, I am finally getting logs from rxfax. Unfortunately, I am still not receiving faxes correctly. Here is the log that was produced. If anyone has any thoughts on what might going, I would greatly appreciate it. I'm actually getting *some* blank faxes too...it seems that I can receive from an older crappy fax machine but not from a newer one like an HP all-in-one...how do you get debugging information so I can possibly help with this problem too? -- Happiness Is Seeing Your Mother-in-law on a Milk Carton. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DLink mgcp phone and CVS HEAD
Hi, I'm playing around with Asterisk and DPH-100M (Dlink mgcp phone) on my debian box. I've got stable version of Asterisk (packaged for debian) working with dlink phone and 7910 from cisco (minimalistic extensions.conf and chan_skinny for 7910) Everything works fine. Now I'm trying to get CVS HEAD working with MGCP. I want to test chan_sccp (http://sourceforge.net/projects/chan-sccp/) and this compiles for head only. I've successfuly compiled CVS version and installed it. Chan_sccp works as I see, but strange things happens to dlink phone. When I pick-up the phone I hear nothing, but after some period I hear fast busy (or how is it called). If I press hook for very short period (flash), than I can hear tone, but unable to dial extensions. Anybody experienced this? May mgcp debug be posted on list? Thanks in advance. -- Alexei Chetroi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and dial-up modems
Title: Asterisk and dial-up modems Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other?
Re: [Asterisk-Users] Playing the invalid extension input
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.
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Re: [Asterisk-Users] Re: Grandstream CFG file generator
Stephen R. Besch wrote: snip This is the install package for the program. Running setup will install the program into Program Files\SachsLab\GSConfigure and put a shortcut in the start menu under Phones. The sources are installed to the application directory in a folder named Source. If you have never installed a VB6 setup package, then wyou will very likely get the following message: Setup cannot continue because some system files are out of date on your system. Click OK if you would like setup to update these files for you now. You will need to restart Windows before you can run setup again. Click cancel to exit setup without updating these system files. You can safely click OK to update these files. Also, during the install process, if any of the files already on your system are newer than those in the package (you will be notified), you should opt to keep the ones already on your system. If by chance you happen to already have Bruce McKinney's Windows type library on your machine, then you should install the one that comes with this package, since it includes some constant definitions that are not in the original. If, in the extremely unlikely event that you have your owm custiomized version of the type library, then you should contact me off list if you want to play with the sources. However, the files doesn't gets updated or something else. The install process allways ends with that out of date message. Looking at you SETUP.LST the following files are missing from my PC: [Bootstrap Files] [EMAIL PROTECTED],$(WinSysPathSysFile),,,7/15/00 12:00:00 AM,101888,6.0.84.50 File2=OK [EMAIL PROTECTED],$(WinSysPathSysFile),$(TLBRegister),,6/3/99 12:00:00 AM,17920,2.40.4275.1 File4=OK [EMAIL PROTECTED],$(WinSysPathSysFile),$(DLLSelfRegister),,3/8/99 12:00:00 AM,164112,5.0.4275.1 [EMAIL PROTECTED],$(WinSysPathSysFile),$(DLLSelfRegister),,4/12/00 12:00:00 AM,598288,2.40.4275.1 File7=OK BR /t ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Do people actually answer questions here?
On Tue, 2004-06-29 at 10:11, Randy Bush wrote: DO NOT USE GROUP REPLY!!! fat effing chance. fix your mail system. see a very large number of threads. but as you seem unable to look up archives:-), try this in your .procmailrc # prevent dupes # :0 Wh: msgid.lock | formail -D 65536 msgid.cache Maybe you need to think about the fact that when you do that, you toss the second copy. The second copy is the one from the mailing list and therefore the one with the mailing list headers that are sorted by. You could just be considerate and do things The Right Way(tm) Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. this is the cult of this is a very complex area. you need to pay us gurus to do it for you. in my 40 years of computing i can not count the fields where i have seen the guru-friendly products and technologies go one of two ways, marginalization and failure or takeover by the massive companies who then marginalize the engineers. i suggest you have another career path planned. This isn't my career path. The problem is either you will have to spend your time and money learning or your will spend your money on someone else to do it for you. I prefer you learn it yourself, but your impatience proves you probably won't. there are reasons there are so many repeated questions on this list. some of the reasons are weakness in the documentation. we can pretend this is not a problem and attack the questioners or those who try to discuss the weaknesses, or we can discuss how we might address them. clearly you have made your choice. Repeat questions in the same day are due to people not spending _ANY_ time on their research. repeat questions in the same month is the same problem. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hold Button on FireFly does not launch MusicOnHold on *?
I'm using version 1.9.1 build 3908 - next problem is that the text messages won't reach by another firefly client - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Randy Bush [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 7:31 PM Subject: [Asterisk-Users] Re: Do people actually answer questions here? DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! Got that... DO NOT USE GROUP REPLY!!! I will get a copy from the list server just fine without a personal copy to me directly. On Tue, 2004-06-29 at 09:39, Randy Bush wrote: we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. Booo whh, use better tools. Find out how to use Mozilla and tabbed browsing. Tabbed browsing can be set up to make a new tab for any link that goes to a different site. It can also be configured to create new tabs for those annoying popups. Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. If you knew how much time and how much money both of my personal fincances and my employers was put into the knowledge I have currently, you would start to understand why I am so annoyed that you don't seem to want to spend the time it takes to learn. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Do people actually answer questions here?
From: Steven Critchfield [EMAIL PROTECTED] To: Randy Bush [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] ROFL! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing the invalid extension input
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Second time in 2 days, What extention are you at exten = i?
If you want to readback and invalid number you will have to first
capture the number with a patternmatch then do something like a macro to
then read it back.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Vonage and Asterisk integration
I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. (*) Examples: Had three lines on two ATAs. Asked if I can moved one of the lines off to a third (new) ATA -- they couldn't do it. Asked if I can move an existing number to a Softphone line. Nope, couldn't do it. Can I make an existing number a virtual number? Nope, can't do. Apparently, they can utilize LNP to move numbers from you CLEC to themselves, but they can't move numbers around inside Vonage. Ba! It cost them two lines and about $45/month in services. -Original Message- From: Jerry Roy [mailto:[EMAIL PROTECTED] Sent: Monday, June 28, 2004 12:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage and Asterisk integration All, I have been thru the archives and all the relevant URL's sent to me. I have sent e-mail to those who have gone before me and are attempting to accomplish the same goal - no one has it working?. Doesn't anyone have a WORKING asterisk pbx that hooks into vonage? Thanks, Jerry Roy 562-305-9545 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sip.c max number of retries
never mind, new server + upgrades on the phones sofware+ latest asterisk cvs = :) - Original Message - From: Robb Meredeth To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 6:59 PM Subject: Re: [Asterisk-Users] chan_sip.c max number of retries One thing I forgot to mention in the first post. I also can call sip-sip on the old server but not on the new server, however I can call from sip-zap on the new. When I dial from one sip to the other I get no ringback on the calling set and the called phone doesn't ring either. After a short time the calling set gives up and goes to voicemail. I'm sure I'm doing something dumb or I've screwed up a config I'm not thinking of right now. I've googled and deja searches, but I'mstumped. I even reloaded theoson the new machine and I get the exact same error. Thanks again, Robb Has anyone who's gotten this message managed to figure it out and fix it? I've been scouring the mailing list for clues but I'm still no closer. I have 2 asterisk servers, old and new. I'm trying to switch to the new server. I am using a 2.4 kernel on the new and a 2.6 on the old. I am running 0.9.0 on the old and my sip phones work fine, on the new Ihave tried 0.9.0, 0.9.1, and current CVS builds. I never see this error on th eold machine and always on the new machine. I have copied all my configs directly and they are plugged into the same switch as each other and the phones. the only difference (aside from hardware) is the kernels (I had to drop back to get zaptel to compile) and the new has an X100pfxo card. The only change I have made to the phones (zultys 4x4's) is to change thetftp server and the SIP Proxy. If anyone can give me a direction to look in or any advice at all, I would appreciate it greatly. Thanks! Robb
RE: [Asterisk-Users] * Busy-Redial ??
On 01:04 AM 6/29/2004, Florian Overkamp wrote: Hi, snip busy, so you hang up and dial *XX12125551212 and hangup again, then * would continue to retry calling the number until either it rings or a timeout is reached, if it rings * then calls back the exten that made the *XX call and bridges the two channels (maybe even with a distinctive ring). If anyone has any suggestions on how to accomplish this please let me know. I have not implemented this, but it should be doable by creating a file in the spool directory. You really just need to tinker around with contexts and actions to do 'the right thing' (tm). An AGI script to generate the spool file would probably do the trick. Doing this using an analog Zap channel x100p, tdm400.. will be problematic, as Asterisk's ability to determine if the line has been answered on an analog zap channel is less than perfect. You could easily implement a feature where you could preform redial on busy when using a channel that provides signalling to indicate the call progress, especially if you didn't mind having the calling channel staying connected. .call files might be able to work, but I still can't think of a method to allow the call to progress until the far end begins to ring BEFORE inviting the originating channel from the first transaction. And finally, what kind of call handling do you want if the far end answers before the originating end picks up? Music on hold, special message, or just ringing? While asterisk is quite flexible, and you can almost always find a way to do something, I suspect to make this work in a way that handles the whole process in an understandable and acceptable manner is going to be quite complex and probably a masterpiece of duct tape and bubblegum. This is a feature that would probably best be designed as an Asterisk application, and yet even then, the analog channels will be problematic. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Playing the invalid extension input
Maybe it is trying to say i as a digit?
You could have an [invalid] context with
[invalid]
exten = _.,1,Saydigits(${EXTEN})
and then include it at the very end of the [default] context (or wherever
you want to use it). That would then pick up anything that drops through. If
you do it any other way it will get sorted to the top and you will have
trouble!
Peter
-Original Message-
From: Isamar Maia [mailto:[EMAIL PROTECTED]
Sent: 29 June 2004 15:54
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Playing the invalid extension input
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
Any suggestions?
Thanks,
Isamar
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[Asterisk-Users] MGCP and call waiting, doesn't work.
Hey guys, can you shead some light on
this?
I will copy my mgcp.conf and post below, but here
is the problem.
I can't get call waiting to work with my MGCP
device. I already have one call going, and I can hear the second call come
in, I flash over to it, but all I get is a dial tone, * puts the 1st call on
mute/hold, but I never get the second, and it terminates. I flash back
over and pick up the first.. Here is some debug info
*CLI set verbose 10 --
Accepting call from '9003796075' to '9003790612' on channel 0/1, span
1 -- Executing Dial("Zap/1-1", "MGCP/aaln/[EMAIL PROTECTED]|30|tr")
in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: -1, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created
in state: Down -- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is
ringing -- Endpoint 'aaln/[EMAIL PROTECTED]' observed
'hd' -- MGCP/aaln/[EMAIL PROTECTED] answered
Zap/1-1 -- Accepting call from '2172026046' to
'9003790612' on channel 0/2, span 1 -- Executing
Dial("Zap/2-1", "MGCP/aaln/[EMAIL PROTECTED]|30|tr")
in new stack -- MGCP mgcp_request(aaln/[EMAIL PROTECTED])
-- MGCP cw: -1, dnd: 0, so: 1, sno: 0 -- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created
in state: Down -- Called aaln/[EMAIL PROTECTED]
-- MGCP/aaln/[EMAIL PROTECTED] is
ringingJun 29 09:46:58 NOTICE[98310]: chan_mgcp.c:2268 handle_response:
Terminating on result 400 from aaln/[EMAIL PROTECTED] == No
one is available to answer at this time -- Endpoint 'aaln/[EMAIL PROTECTED]' observed
'hf' -- Swapping 1 for 0 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 1 on aaln/[EMAIL PROTECTED]
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created
in state: Down -- Endpoint 'aaln/[EMAIL PROTECTED]' observed
'hf' -- Swapping 0 for 1 on aaln/[EMAIL PROTECTED]
-- We didn't make one of the calls FLIPFLOP 0 and 1 on aaln/[EMAIL PROTECTED]
-- MGCP Muting 0 on aaln/[EMAIL PROTECTED]Jun 29 09:47:08
WARNING[278545]: pbx.c:1892 ast_pbx_run: Timeout, but no rule 't' in context
'pstn-in' -- Hungup 'Zap/2-1' --
Endpoint 'aaln/[EMAIL PROTECTED]'
observed 'hu' == Spawn extension (pstn-in, 9003790612, 1) exited
non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1'
Here is my MGCP.conf
[10.252.240.2]context=mainhost=10.252.240.2nat=nocallerid
= "Kevin Osterbur"
9003790610callwaiting=yesthreewaycalling=notransfer=nocancallforward=nocanreinvite=noline
= aaln/1context=mainhost=10.252.240.2nat=nocallerid = "Duane
Cox"
9003790612callwaiting=yesthreewaycalling=notransfer=nocancallforward=nocanreinvite=noline
= aaln/2
Any ideas?
RE: [Asterisk-Users] Asterisk and dial-up modems
Title: Asterisk and dial-up modems Look at the ZapRAS 'show application ZapRAS' this only work w/a PRI. TL -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of John VogelSent: Tuesday, June 29, 2004 11:24 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and dial-up modems Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other?
Re: [Asterisk-Users] Vonage and Asterisk integration
Jay Milk wrote: I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. I never worked with vonage, is there tech support that bad? -- Regards, Steve Kalcevich, This electronic message contains information from Primus Telecommunications Canada Inc. (PRIMUS) , which may be legally privileged and confidential. The information is intended to be for the use of the individual(s) or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this electronic message in error, please notify us by telephone or e-mail (to the number or address above) immediately. Any views, opinions or advice expressed in this electronic message are not necessarily the views, opinions or advice of PRIMUS. It is the responsibility of the recipient to ensure that any attachments are virus free and PRIMUS bears no responsibility for any loss or damage arising in any way from the use thereof.The term PRIMUS includes its affiliates. Pour la version en français de ce message, veuillez voir http://www.primustel.ca/fr/legal/cs.htm begin:vcard fn:Steven Kalcevich, CCNA, CCDA n:Kalcevich;Steven org:Primus Canada;Commercial Sales adr:Suite 400;;5343 Dundas St W;Toronto;ON;M9B 6K5;Canada email;internet:[EMAIL PROTECTED] title:Account Executive tel;work:416-207-4613 tel;fax:1-800-861-3035 x-mozilla-html:FALSE url:http://www.primustel.ca/ version:2.1 end:vcard
[Asterisk-Users] channel.c:1508 ast_set_read_format: Unable to find a path from ULAW to UNKN
Hello all- This is probably easy, but I am trying to figure out why I cannot use app record. In fact, the problem seems to extend to when using a sip soft phone also. I always get this error, and haven't been able to find much about it. I have a number coming in from nufone. I have tryied setting the codec and also leaving it out. See config below: ; answer the line exten = number,1,SetVar(SIP_CODEC=ulaw) exten = number,2,Wait(2) exten = number,3,Answer exten = number,4,ResponseTimeout(15) exten = number,5,Background(pls-wait-connect-call) I then have a recording section: exten = 205,1,Wait(2) exten = 205,2,Playback(all-your-base) ;exten = 205,2,Playback(/usr/local/asterisk/var/lib/asterisk/sounds/hello-recording) exten = 205,3,Record(/usr/local/asterisk/var/lib/asterisk/sounds/hello-recording:gsm ,2) exten = 205,4,Wait(1) exten = 205,5,Playback(/usr/local/asterisk/var/lib/asterisk/sounds/hello-recording) exten = 205,6,Wait(1) exten = 205,7,Hangup I always get the following error: channel.c:1508 ast_set_read_format: Unable to find a path from ULAW to UNKN In iax.conf I have disallow=all allow=ulaw I have tried replacing all references to ulaw with gsm. Same results. I have used cvs head and also 0.9.0. Ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage and Asterisk integration
Vonage does not allow any other device other than their own to be hooked up to their system, period. There are whole bunch of service providers who allow you to hook-up your own device. So why split hairs, use someone else other than Vonage. Their is nothing extraordinary about Vonage, except they have some advertising dollars. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jay Milk Sent: Tuesday, June 29, 2004 8:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Vonage and Asterisk integration I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. (*) Examples: Had three lines on two ATAs. Asked if I can moved one of the lines off to a third (new) ATA -- they couldn't do it. Asked if I can move an existing number to a Softphone line. Nope, couldn't do it. Can I make an existing number a virtual number? Nope, can't do. Apparently, they can utilize LNP to move numbers from you CLEC to themselves, but they can't move numbers around inside Vonage. Ba! It cost them two lines and about $45/month in services. -Original Message- From: Jerry Roy [mailto:[EMAIL PROTECTED] Sent: Monday, June 28, 2004 12:51 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Vonage and Asterisk integration All, I have been thru the archives and all the relevant URL's sent to me. I have sent e-mail to those who have gone before me and are attempting to accomplish the same goal - no one has it working?. Doesn't anyone have a WORKING asterisk pbx that hooks into vonage? Thanks, Jerry Roy 562-305-9545 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] nat problem
hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call from sip gateway to asterisk : no problem outgoing call from asterisk to sip gateway : no sound no sip error if i put a phone on public ip, incoming and outgoing call work fine i haven't make any port translation since phone and asterisk are on the same private subnet. phone register on the private ip of asterisk i have zaphfc connected to isdn and incoming and outgoing call work fine (with phone on public or private ip) any idea what's wrong or try to fix this problem ? thanks for help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modifyed Prepaid Aplication!
Hello I have succesfully installed app_prepaidCID and populated the database. I can send calls, but the only problem is that after call finish it does not update the billing ballance on card. can any one help me about this? Best Regards Hekuran Doli ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Blank faxes with RxFAX
I wasn't able to get debugging information the first time around either. After pulling the latest asterisk from CVS, I was able to build and see debugging information when I started asterisk to test using asterisk -vvgc. But I noticed today that I do not get the same debugging information when I started asterisk using safe_asterisk. So, I don't know that rebuilding asterisk did any good. If you started asterik using safe_asterisk, I would shut it down, restart using asterisk -vvgc from a shell prompt and it may give you debugging information from the CLI. Hope that helps. Patrick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Hirsch Sent: Tuesday, June 29, 2004 11:13 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Blank faxes with RxFAX Patrick J. Conroy wrote: Hello All, I have finally pulled CVS HEAD and built it with app_rxfax and app_txfax to try to solve the problem that I was having with blank faxes. Fortunately, I am finally getting logs from rxfax. Unfortunately, I am still not receiving faxes correctly. Here is the log that was produced. If anyone has any thoughts on what might going, I would greatly appreciate it. I'm actually getting *some* blank faxes too...it seems that I can receive from an older crappy fax machine but not from a newer one like an HP all-in-one...how do you get debugging information so I can possibly help with this problem too? -- Happiness Is Seeing Your Mother-in-law on a Milk Carton. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content, and is believed to be clean. -- This message has been scanned for viruses and dangerous content, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Complaining Emails
Hi Everyone, I'm one of those newbie users that really doesn't know what is going on. And in stark contrast to the posts you may read, I do actually think that when you post on this list, you get a reply. In fact, I know you do. I have had a quite nice run setting up asterisk, when I've had problems I've looked at the Wiki or Google, and then when I can't find an answer, then I'll post - and everyone has been very helpful in getting my system running. Of course - I've made my fair share of mistakes. Not all lists work quite the same way... and at time's I've been a bit over eager, and over zealous. And I'm sure I've had my posts simply marked as read or deleted by many people - some without opening. But I have a strategy for dealing with the people who see fit to complain: Don't reply. It's that simple. If some people want to insult the intelligence or care of the people on this list, there is little we can do about it - except ignore them. Then, when they complain about people not replying, we can simply say to ourselves - it's there own fault. And it benefits those of us who do use the list regularly, because there is fewer messages to sift through. The other issue that was mentioned was that of people not bothering to do their research before hand. In reality - the mailing list does exist to allow people to provide personalized answers to other people's questions. And a lot of these will be duplicate. I don't know - but I think the best strategy is to answer the question, and then point people to the Wiki as the place to look. I don't want the Steve's, Kevin's or Eric's to leave this list. (Those are plural apostrophes btw). We all appreciate their help - hey - I appreciate their help. Maybe one day, I'll be fortunate enough to know as much as they do about this software, and be able to provide the same level of help to other people. Let's not spoil what we have. Just my 2 cents... Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Playing the invalid extension input
Eric Wieling wrote:
On Tue, 2004-06-29 at 09:53, Isamar Maia wrote:
I'm trying to do the following:
exten = i,1,Saydigits(${EXTEN})
My intention is to play the invalid input to the user, but it doesn't
work.
At that pint ${EXTEN} is i. Try using ${INVALID_EXTEN}
Eric,
Thank you, I've added that variable to the Wiki.
/O
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RE: [Asterisk-Users] Hold Button on FireFly does not launch MusicOnHold on *?
You replied to a message with the subject of: Re: Do people actually answer questions here? And then changed the subject and started typing. This has wreaked havoc on everybody's threaded readers, and made your question impossible to reply to. You need to start a new message in your mail app and start from scratch. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shabanip Sent: Tuesday, June 29, 2004 10:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hold Button on FireFly does not launch MusicOnHold on *? I'm using version 1.9.1 build 3908 - next problem is that the text messages won't reach by another firefly client - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: Randy Bush [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 7:31 PM Subject: [Asterisk-Users] Re: Do people actually answer questions here? DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! DO NOT USE GROUP REPLY!!! Got that... DO NOT USE GROUP REPLY!!! I will get a copy from the list server just fine without a personal copy to me directly. On Tue, 2004-06-29 at 09:39, Randy Bush wrote: we point people to the wiki problem is that wikiware search sucks caterpillar snot, and this particular wiki is a bit light on content and heavy on links. one can spend massive time following links seemingly relevant to a subject and never get to actual content about it. often google yields better results. Booo whh, use better tools. Find out how to use Mozilla and tabbed browsing. Tabbed browsing can be set up to make a new tab for any link that goes to a different site. It can also be configured to create new tabs for those annoying popups. Plain and simple, this is a complex subject matter, it WILL take time to learn. There may not be quick answers for you unless it is to hire someone else to do it for you. This is a fact of life. Learning is not always easy. Many of us have spent a LARGE sum of money on tools and documentation, and then there is the amount of time. If you knew how much time and how much money both of my personal fincances and my employers was put into the knowledge I have currently, you would start to understand why I am so annoyed that you don't seem to want to spend the time it takes to learn. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do people actually answer questions here?
Leif Madsen wrote: On Tue, 29 Jun 2004 11:08:43 +1000, Jean-Yves Avenard [EMAIL PROTECTED] wrote: I have to admit I'm rather disappointed with Asterisk, information is probably available but very hard to find ; it seems to be limited to a few privileged people for whom their job is setting up VoIP system Based on your statement, I would presume that you have never even attempted to search for documentation. I can think of at least 3 excellent resources: http://www.voip-info.org http://www.fnords.org/~eric/asterisk/ http://www.asteriskdocs.org Plus using site:lists.digium.com and site:voip-info.org in Google is an excellent resource. Don't forget the most important link of all at the bottom of every email. The unsubscribe link! -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dead Budgetone-101?
On Sun, 27 Jun 2004, Max Lock wrote: Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank screen, although the keypad lights blink occasionally. After that I tried to go back to the 1.0.4.55 firmware I was running previously, but the tftp transfer no longer seems to work, the phone sends a tftp request, but doesn't seem to get the files?! Has anyone else seen this? is there a way to access the phone via the internal PCB edge connector? -Cheers Max Call GrandStream technical support. They can help you UnBrick your phone, provided the Ethernet port still works. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Customized Call Parking
On 29 Jun 2004, Adnan Shah wrote: Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Put up a bounty on it of a reasonable amount, and I'm sure that someone could code up an app to do it. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T100P-E100P circuit board differences
Hi- Perhaps someone with an E100P in hand can answer this: I just received an E100P from Digium (I normally buy quad boards) I noticed that the circuit board says T100P on it, and I assume that the T100P and E100P both use the same circuit board. Can someone please confirm that their E100P says T100P on the artwork? Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ldap-lookup
hi, is there a ldap-lookup for asterisk ?? i am seraching for asterisk app which i can give a name and a phone-type (sip, iax, cellphone, work, home.) and get phonenumber back. And where i can give a number and get the name. Is there anything like that? bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting Asterisk to automatically dialout
Hi, I'm trying to get asterisk to auto-dail out. I created a *.call file with the the top of it being Channel: Zap/1/2609944, which should have connected to Zap channel 1 and dial out to 2609944, but It did not do so, asterisk would say a call was completed to Zap/1/2609944 but I never heard that phone ring. So I tried just putting Channel: Zap/1 at the top of the call file so it would connect to Zap channel 1, then in the *.call file connect it to an outgoing context in extensions.conf which looked like: [outgoing] exten = s,1,Wait(1) exten = s,2,Dial(Zap/1/2609944) exten = s,3,Wait(2) exten = s,4,Playback(soundfile) exten = s,5,Hangup But when it ran this, asterisk told me it was unable to create a channel of type Zap, but then that a call was still completed to Zap/1. I've read everything about auto-dialout on voip-info.org and read digium faqs and everything and have been unable to find a solution. If someone out there has had a similar problem and figured it out or knows what might be wrong with what I'm trying to do it would be greatly appreciated if you could help me out. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] linux kernel 2.6.6
On Tue, 2004-06-29 at 05:10, Kevin Walsh wrote: I'm using 2.6.7-gentoo-r6 and it works very well. I assume you're using the latest Zaptel and Asterisk from CVS. Do you have CVS ebuilds? That would make it a lot easier for us Gentoo folks. Thanks! -- Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ruggedised IP Phone
- Original Message - From: Matt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 12:42 AM Subject: [Asterisk-Users] Ruggedised IP Phone Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. See: http://www.voip-info.org/wiki-Asterisk+phone+door Please add any new info you find. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage and Asterisk integration
Like I said, they just seem to be lazy and/or badly organized. If they can do LNP, why can't they change a hardline into a softphone, break one number out onto a different ATA, etc? I basically laid it out for them, saying If you can't move my 2nd line from this ATA to a new ATA, then I'll need to cancel that line... I no longer have that line. Not being able to something this simple cost them over $500/year from me... I wonder how many other Vonage users will drop them because of such things. -Original Message- From: Steve Kalcevich [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 11:01 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Vonage and Asterisk integration Jay Milk wrote: I do. I decided not to bother with Vonage's sub-par and unmotivated customer service(*) and plugged my ATA186 into an FXO port. I never worked with vonage, is there tech support that bad? -- Regards, Steve Kalcevich, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP600 stops to send/receive calls
Jorge, That sounds strange to me. I have 12 IP 600s running without any of the problems that you are having. My first guess would be that they are configured wrong. Is the phone registering with asterisk? Is the phone dowloading it's config files from the FTP server? If you want to post your phone1.cfg and sip.conf, I can try to see if there is anything wrong with it. -Tor Jorge Mendoza wrote: Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls. After a reboot is works fine again. We have a * box with many BT101 and softphones working for months without any problem. I'm missing something? it is a bad config file? or it is a phone bug? Thank You for your time. Jorge Mendoza ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play Music on hold until a ZAP channel frees up.
[answeringsvc]
exten = 0,1,Wait,1
exten = 0,2,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},r)
exten = 0,3,Dial(ZAP/g1/7135551212,${RING_TIMEOUT},mr)
exten = 0,103,Goto(0,3)
exten = 0,104,Goto(0,3)
This should call 713-555-1212. If there are no ZAP lines available it
should kick back around and play music on hold until a zap line is
available, correct? I'd like the music-on-hold to be continuous until
the Zap line answers.
TIA,
--
Daniel Jimenez djimenez[at]pobox[dot]com
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RE: [Asterisk-Users] Vonage and Asterisk integration
On Tue, 2004-06-29 at 11:06, Doug Harris wrote: Vonage does not allow any other device other than their own to be hooked up to their system, period. There are whole bunch of service providers who allow you to hook-up your own device. So why split hairs, use someone else other than Vonage. Their is nothing extraordinary about Vonage, except they have some advertising dollars. And unlimited calling plans (not something most people will care about if they do the math), and one of the best coverage areas when it comes to DIDs. Packet8 is similar to Vonage in the good and the bad ways. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting Asterisk to automatically dialout
On Tue, 29 Jun 2004, Andrew Elchuk wrote: I'm trying to get asterisk to auto-dail out. I created a *.call file did you create the file in /var/spool/asterisk/outgoing/, or did you create it elsewhere and then move it to that directory? The docs mention that if the file is created in the outgoing directory, * may read the file before you've finished writing it. But if you create the file elsewhere and then move it, the entire contents are guaranteed to be in the filesystem when * finds it. This might explain the difficulty you're having.. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Customized Call Parking
This sounds like it should be relatively simple to do in theory. Couldn't you just create specific extensions that set the MusicOnHold context (to play your different announcements) then transfers the call to the parkext in parking.conf? However, what do you want to happen when the announcement is finished? If you park the call there is no way to know when the announcement is finished so you can return to the call later (if thats what you want to do). Perhaps just creating a sort of menu you transfer the calls to, that plays the announcement of your choice, then when its done, it could automatically transfer the call back to the originating extension, or park the call, or put it on hold, or hang up, etc... I think what your looking for is definitely possible without writing a custom application for it. On Tue, 2004-06-29 at 15:06 +0600, Adnan Shah wrote: Hi ! I need a solution to park incoming calls to an extension of my choice where a special announcement is played, park subsequent calls to specific pools so that they listen to announcements of my choice. any ideas ? Shah. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Benoit [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nat problem
Webn1 a écrit : hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call from sip gateway to asterisk : no problem outgoing call from asterisk to sip gateway : no sound no sip error if i put a phone on public ip, incoming and outgoing call work fine i haven't make any port translation since phone and asterisk are on the same private subnet. phone register on the private ip of asterisk i have zaphfc connected to isdn and incoming and outgoing call work fine (with phone on public or private ip) any idea what's wrong or try to fix this problem ? codec? thanks for help -- Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
