[Asterisk-Users] 7940
I have a new 7940 I have set-up the network And tried to tftp SIP ver. 2.1 And ever time it boots and starts the tftp download the 7940 reboots Any input welcome Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and latest Debian package
At 18:45 8-10-2003 +0200, you wrote: Hi capi users :-) you might also want to try chan_capi 0.3.0 which is already in the downloads directory but not linked on the page. The option echosquelch=1 now finally works. Yeah, I found 0.3.0 recently and installed it, seems to be working fine. What is echosquelch supposed to do ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 5 second latency sip to oh323
hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred the scenario is this: sip-asterisk-h323:operator (who then transfers the call) h323:destination --audio path 5-second latency audio path ok--- here is the output of the "show channels" H323:19742 (voip s 1 ) Up Bridged Call SIP/kelvin-6952SIP/kelvin-6952 (voip 2010 1 ) Up Dial OH323/H323:[EMAIL PROTECTED]|25|mt the problem only exists in transferred calls any infowould be appreciated thanks =) ~kelvin
[Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in a centralized routing arbiter model 9) Speech recognition support Care to add your own unicorns to the list? I make no judgement nor do I cast aspersions on any of these items, but I seem to recall seeing comments about I'm working on... or It would be really great if... on all of these without seeing real evidence on any of them other than talk. The only well-remembered myth I can say for certain that has been dispelled is the SCCP channel driver, and that has been moved out of Loch Ness status to peer-reviewed status. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7940/7960 phone and conference calling ?
I am guessing you are running without reinvite's, I'm running with reinvite's with latest CVS release and 79x0 phones without any issues with conferencing... -Original Message- From: Adam Rothschild [mailto:[EMAIL PROTECTED] Sent: 08 October 2003 15:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7940/7960 phone and conference calling? Hello, Anyone else having problems with the Cisco 7940/7960 (5.3 firmware) and the latest CVS build, placing conference calls from the phone? I've noticed the party on the Cisco phone's side will sound very garbled, and delayed by several seconds. I haven't begun troubleshooting yet, though I'm able to reproduce this easily... Thanks in advance, -a ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * 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
Re: [Asterisk-Users] 7940/60 TFTP Problem
John Todd wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Babak Some hints which may get you going: http://www.loligo.com/asterisk/Cisco/79xx/upgrading.79xx.phones Re-name the files on your TFTP server to shorter names; I know I used that trick at least a few times in the past. JT OK, As per John Todd's suggestion I started playing around with file names trying a variety until I got the gollowing name to work without genertating malformed packets: P0S3-05-.bin However, now it starts the TFTP process and after packet 769 of the TFTP ethereal gives me: Error Code, Code: Disk full or allocation exceeded, Message The process is in a loop over and over and over again. Babak -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 Note: please post follow-ups on the bottom of your message; it keeps things in chronological order. I had a phone do the same thing, and after about 100 reboots it magically worked. I have no idea why; the person who was working on it simply gave up and let it sit on his desk and cycle for (1? 3? 5? days) and he came back and it was working. Try loading one of the other images first, perhaps one of the smaller ones (3.2.2) and see if that solves any problems. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 TFTP Problem
All I get is Version Error When trying to tftp Any ideas ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, 9 October 2003 5:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem John Todd wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Babak Some hints which may get you going: http://www.loligo.com/asterisk/Cisco/79xx/upgrading.79xx.phones Re-name the files on your TFTP server to shorter names; I know I used that trick at least a few times in the past. JT OK, As per John Todd's suggestion I started playing around with file names trying a variety until I got the gollowing name to work without genertating malformed packets: P0S3-05-.bin However, now it starts the TFTP process and after packet 769 of the TFTP ethereal gives me: Error Code, Code: Disk full or allocation exceeded, Message The process is in a loop over and over and over again. Babak -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 Note: please post follow-ups on the bottom of your message; it keeps things in chronological order. I had a phone do the same thing, and after about 100 reboots it magically worked. I have no idea why; the person who was working on it simply gave up and let it sit on his desk and cycle for (1? 3? 5? days) and he came back and it was working. Try loading one of the other images first, perhaps one of the smaller ones (3.2.2) and see if that solves any problems. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
wan't to add DS3 and SS7 to that ? also licensed g723.1 and working g729 softfax and softmodem that's what comes to my mind on the spot ... On Thursday 09 October 2003 9:51 am, John Todd wrote: Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in a centralized routing arbiter model 9) Speech recognition support Care to add your own unicorns to the list? I make no judgement nor do I cast aspersions on any of these items, but I seem to recall seeing comments about I'm working on... or It would be really great if... on all of these without seeing real evidence on any of them other than talk. The only well-remembered myth I can say for certain that has been dispelled is the SCCP channel driver, and that has been moved out of Loch Ness status to peer-reviewed status. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX
Hello All, Is it possible to make asterisk to do authetication of IAX client through database (mysql, etc) instead of creating all the client username in iax.conf? How hard is to implementthe feature i describe above? We plan to use IAX as part of our VOIP infrastructure mainly because it penetrate NAT/firewall with ease. Foong
Re: [Asterisk-Users] 7940/60 TFTP Problem
Had those same problems with some 7960's but not with others. As previously mentioned (below and by others on the list over the last year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x and somewhere around v4.x remove all the comment lines in SIPDefault. It will work. The problem has been observed by many. Since none of us have 7960 source code, we can only guess at what the root problem is. Or, you can put the phones under Cisco maintenance, call the TAC center, and they'll tell you to do the same thing. Thanks to all that have responsed on this issue. To clarify, I have been able to successfully upgrade at least 4 phones. 2 are giving me the malformed packet problem. They were both running MGCP 3.3 code by the way. The phone goes through it's standard boot sequence, then looks for the file OS79XX.TXT which it sees and downloads properly. This file only contains the image version I want the phoen to have: P0S3-05-3-00 It then tries to download the file but iserts some garbage characters in the name: P0S3-05-3-00Garbage.bin at which point it does not upgrade since the file name contains garbage characters. This is the second phone I have seen do this same exact thing. The first was a 7940, this is now a 7960. So in summary the process I have been using has been successful except with two phone which are doing the same exact thing. Thanks again for the attention to this matter. Babak Rich Adamson wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Without specific error codes, packet traces, etc, we can only guess. Some real options include: a) not all tftp servers are the same. Over about 15 years of experience, many tftp servers have an issue with the last packet and how to close the session. Might research the exact software you're using. b) the cisco phones seem to have an issue with the comment lines in the SIPDefault file (on the tftp server). Remove every one and test again. This seems to be highly dependent on the exact versions of SIP code implemented (in sequence) from v2.x upward. Some phones seem to have the problem while others don't. Since we don't know what version of code was in the phone when you received it, best guess is to boot to v2.x, then 3.x, then 4.x, then 5.x and I'll bet (at least a beer) the problem disappears. Why? don't know, but lots of similar comments on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 TFTP Problem
Well I eventually got the 7940 loaded Now does anyone have quick fix to get it to work with asterisk Tar in advance Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, 9 October 2003 9:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem Had those same problems with some 7960's but not with others. As previously mentioned (below and by others on the list over the last year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x and somewhere around v4.x remove all the comment lines in SIPDefault. It will work. The problem has been observed by many. Since none of us have 7960 source code, we can only guess at what the root problem is. Or, you can put the phones under Cisco maintenance, call the TAC center, and they'll tell you to do the same thing. Thanks to all that have responsed on this issue. To clarify, I have been able to successfully upgrade at least 4 phones. 2 are giving me the malformed packet problem. They were both running MGCP 3.3 code by the way. The phone goes through it's standard boot sequence, then looks for the file OS79XX.TXT which it sees and downloads properly. This file only contains the image version I want the phoen to have: P0S3-05-3-00 It then tries to download the file but iserts some garbage characters in the name: P0S3-05-3-00Garbage.bin at which point it does not upgrade since the file name contains garbage characters. This is the second phone I have seen do this same exact thing. The first was a 7940, this is now a 7960. So in summary the process I have been using has been successful except with two phone which are doing the same exact thing. Thanks again for the attention to this matter. Babak Rich Adamson wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Without specific error codes, packet traces, etc, we can only guess. Some real options include: a) not all tftp servers are the same. Over about 15 years of experience, many tftp servers have an issue with the last packet and how to close the session. Might research the exact software you're using. b) the cisco phones seem to have an issue with the comment lines in the SIPDefault file (on the tftp server). Remove every one and test again. This seems to be highly dependent on the exact versions of SIP code implemented (in sequence) from v2.x upward. Some phones seem to have the problem while others don't. Since we don't know what version of code was in the phone when you received it, best guess is to boot to v2.x, then 3.x, then 4.x, then 5.x and I'll bet (at least a beer) the problem disappears. Why? don't know, but lots of similar comments on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] chan_capi and latest Debian package
Hi, echosquelch=1 enables Petr Michalek's echo canceler, which compares RX and TX volumes and mutes the RX in an echo condition. regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk Am Don, 2003-10-09 um 09.09 schrieb Florian Overkamp: At 18:45 8-10-2003 +0200, you wrote: Hi capi users :-) you might also want to try chan_capi 0.3.0 which is already in the downloads directory but not linked on the page. The option echosquelch=1 now finally works. Yeah, I found 0.3.0 recently and installed it, seems to be working fine. What is echosquelch supposed to do ? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
wan't to add DS3 and SS7 to that ? I dunno; I've provisioned at least a half dozen DS3s and physically seen one SS7... :-) Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbe Echo problem
I have just (last month) brought up * with ISDN soft phones. I am using Workstation-SIP or iax-*-isdn4linux-hisax-EICON Diva ISDN (not Pro)-uk(bt)isdn lines I am currently trying SIP clients - the last is an evaluation of SJphone, but this problem does not seem to depend on the Workstation end Outgoing calls leave unacceptable levels of echo, noticable by the outside party, but almost none from the 'inside'. Does anyone know if echo cancelation is applied by carriers at their analog/isdn bridges? Or do we have to provide it on the back end of the isdn? I'm assuming that my echo is not a local problem, but ... anyone any ideas? Dave Kitchen - InSync Technology Ltd ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] my phone shows asterisk
Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] real billing time for a call
yes, you're right, i tried to put a ast_cdr_answer when queue makes the ast_cdr_setdstchan, but it didn't work, or maybe, apps like backgound and playback should not issue an ast_cdr_answer, so it is ussued only when the answer is actually answered by someone. is you used the queue app, don't you also see a 20 sec event for that ? [EMAIL PROTECTED] wrote: hello, I am working with asterisk and looking for some stats about operators, then i've found that there is no real time of the call in asterisk when i use an autoattend context. looking into the cdr.c i can see that applications can call a ¨set destination¨ or something to update the CDR record so you can know the real destination of the call, but i can't found something to make the apps(queue,dial, etc.) to update also the real time of the answer for that call. When the exten,s,1, is executed the answer time is setted and it remains that way, so if for example, a person dials a PBX, the autoattend starts telling him about the menu and the extensions and the person just dial an extension, lets say it took him 15 secs, then the Dial app is executed, for example it could be a queue app, and the extension start ringing, for lets say 5 sec and we have 20 secs so far and no real answer for that call, when anotehr person actually answers the call and they talk about 20 secs, the CDR will tell me that the specific call i'm talking about had 40 secs with 40 secs billables, when the real thing is that it was 20 secs what the real call last, i mean for real when a person actually gives attention to the caller. anyone has opinions? i think it could be very usefull, cause sometimes you need to know, for example, if operators are answering the calls for real or not, or if they just let it ring. with actual statistics.. i can't know that. thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in a centralized routing arbiter model 9) Speech recognition support 10) Database abstraction module for ...CDR, SQL Dial Plan, DBGet/Put, * config files 11) SoftFaxModem 12) SS7 Signaling 13) WEB Interface for Users/Admin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
On Thu, 2003-10-09 at 05:14, Klaus-Peter Junghanns wrote: dont forget generic voice modem support. or even better chan_modoss or chan_modalsa the combination of an external modem (for signalling) and a sound card :) This sounds like your wish list, not something anyone said they would work on. Am Don, 2003-10-09 um 11.54 schrieb Michael Bielicki: wan't to add DS3 and SS7 to that ? also licensed g723.1 and working g729 softfax and softmodem that's what comes to my mind on the spot ... On Thursday 09 October 2003 9:51 am, John Todd wrote: Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in a centralized routing arbiter model 9) Speech recognition support Care to add your own unicorns to the list? I make no judgement nor do I cast aspersions on any of these items, but I seem to recall seeing comments about I'm working on... or It would be really great if... on all of these without seeing real evidence on any of them other than talk. The only well-remembered myth I can say for certain that has been dispelled is the SCCP channel driver, and that has been moved out of Loch Ness status to peer-reviewed status. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] real billing time for a call
Is there any way that you could trigger events based upon actual pickup of calls and hangups of lines in ALL cases(parked calls, queued calls, calls triggerd by .call queue files)? It seems like Asterisk needs something a little lower level to allow for this, is it even possible? MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] real billing time for a call yes, you're right, i tried to put a ast_cdr_answer when queue makes the ast_cdr_setdstchan, but it didn't work, or maybe, apps like backgound and playback should not issue an ast_cdr_answer, so it is ussued only when the answer is actually answered by someone. is you used the queue app, don't you also see a 20 sec event for that ? [EMAIL PROTECTED] wrote: hello, I am working with asterisk and looking for some stats about operators, then i've found that there is no real time of the call in asterisk when i use an autoattend context. looking into the cdr.c i can see that applications can call a ¨set destination¨ or something to update the CDR record so you can know the real destination of the call, but i can't found something to make the apps(queue,dial, etc.) to update also the real time of the answer for that call. When the exten,s,1, is executed the answer time is setted and it remains that way, so if for example, a person dials a PBX, the autoattend starts telling him about the menu and the extensions and the person just dial an extension, lets say it took him 15 secs, then the Dial app is executed, for example it could be a queue app, and the extension start ringing, for lets say 5 sec and we have 20 secs so far and no real answer for that call, when anotehr person actually answers the call and they talk about 20 secs, the CDR will tell me that the specific call i'm talking about had 40 secs with 40 secs billables, when the real thing is that it was 20 secs what the real call last, i mean for real when a person actually gives attention to the caller. anyone has opinions? i think it could be very usefull, cause sometimes you need to know, for example, if operators are answering the calls for real or not, or if they just let it ring. with actual statistics.. i can't know that. thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] real billing time for a call
really, for that, CDR needs to be rewritten in some parts, cause one thing you could use is to know the full path of a call, based on an identifier or something, so you can now the cal last 10 seconds on the prompt, 15 seconds on a queue, the 20 seconds talking, after that was parked for 10 seconds and then 5 secs after picked up and transfered and of ourse actual call last 120 secs. I mean a CDR entrance for each application.. don't yopu think? but that work is quite hard... i think.. What do you say mark? cita quien=mattf Is there any way that you could trigger events based upon actual pickup of calls and hangups of lines in ALL cases(parked calls, queued calls, calls triggerd by .call queue files)? It seems like Asterisk needs something a little lower level to allow for this, is it even possible? MATT--- -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, October 09, 2003 8:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] real billing time for a call yes, you're right, i tried to put a ast_cdr_answer when queue makes the ast_cdr_setdstchan, but it didn't work, or maybe, apps like backgound and playback should not issue an ast_cdr_answer, so it is ussued only when the answer is actually answered by someone. is you used the queue app, don't you also see a 20 sec event for that ? [EMAIL PROTECTED] wrote: hello, I am working with asterisk and looking for some stats about operators, then i've found that there is no real time of the call in asterisk when i use an autoattend context. looking into the cdr.c i can see that applications can call a ¨set destination¨ or something to update the CDR record so you can know the real destination of the call, but i can't found something to make the apps(queue,dial, etc.) to update also the real time of the answer for that call. When the exten,s,1, is executed the answer time is setted and it remains that way, so if for example, a person dials a PBX, the autoattend starts telling him about the menu and the extensions and the person just dial an extension, lets say it took him 15 secs, then the Dial app is executed, for example it could be a queue app, and the extension start ringing, for lets say 5 sec and we have 20 secs so far and no real answer for that call, when anotehr person actually answers the call and they talk about 20 secs, the CDR will tell me that the specific call i'm talking about had 40 secs with 40 secs billables, when the real thing is that it was 20 secs what the real call last, i mean for real when a person actually gives attention to the caller. anyone has opinions? i think it could be very usefull, cause sometimes you need to know, for example, if operators are answering the calls for real or not, or if they just let it ring. with actual statistics.. i can't know that. thanks in advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sistemas - ANALITCA - MD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7940/60 TFTP Problem
Rich, Thank you for your response. I have tried to do that. Unfortunately the oldes version of code available on the Cisco site is 3.2. The current code rev on the phone is 3.3 MGCP. Unfortunately I get the same results. I start with version 3.2 - Did not work, then tried 4.4 and that did not work. We all already know 5.3 has not been working. Would someone have a 2.2 SIP that I could try to keep in accordance to Rich's methodology? Babak Rich Adamson wrote: Had those same problems with some 7960's but not with others. As previously mentioned (below and by others on the list over the last year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x and somewhere around v4.x remove all the comment lines in SIPDefault. It will work. The problem has been observed by many. Since none of us have 7960 source code, we can only guess at what the root problem is. Or, you can put the phones under Cisco maintenance, call the TAC center, and they'll tell you to do the same thing. Thanks to all that have responsed on this issue. To clarify, I have been able to successfully upgrade at least 4 phones. 2 are giving me the malformed packet problem. They were both running MGCP 3.3 code by the way. The phone goes through it's standard boot sequence, then looks for the file OS79XX.TXT which it sees and downloads properly. This file only contains the image version I want the phoen to have: P0S3-05-3-00 It then tries to download the file but iserts some garbage characters in the name: P0S3-05-3-00Garbage.bin at which point it does not upgrade since the file name contains garbage characters. This is the second phone I have seen do this same exact thing. The first was a 7940, this is now a 7960. So in summary the process I have been using has been successful except with two phone which are doing the same exact thing. Thanks again for the attention to this matter. Babak Rich Adamson wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Without specific error codes, packet traces, etc, we can only guess. Some real options include: a) not all tftp servers are the same. Over about 15 years of experience, many tftp servers have an issue with the last packet and how to close the session. Might research the exact software you're using. b) the cisco phones seem to have an issue with the comment lines in the SIPDefault file (on the tftp server). Remove every one and test again. This seems to be highly dependent on the exact versions of SIP code implemented (in sequence) from v2.x upward. Some phones seem to have the problem while others don't. Since we don't know what version of code was in the phone when you received it, best guess is to boot to v2.x, then 3.x, then 4.x, then 5.x and I'll bet (at least a beer) the problem disappears. Why? don't know, but lots of similar comments on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message-
Re: [Asterisk-Users] 5 second latency sip to oh323
How do you transfer the call? Michael. Kelvin Chua wrote: hi guys, i'm using sept 30 cvs and oh323 5.5 i'm having 5 second latecy(on only 1 audio path) when a call is transferred the scenario is this: sip-asterisk-h323:operator (who then transfers the call) h323:destination --audio path 5-second latency audio path ok--- here is the output of the show channels H323:19742 (voip s1 ) Up Bridged Call SIP/kelvin-6952 SIP/kelvin-6952 (voip 2010 1 ) Up Dial OH323/H323:[EMAIL PROTECTED]|25|mt the problem only exists in transferred calls any info would be appreciated thanks =) ~kelvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 TFTP Problem
Mick, Can you please provide more detail on specifically what you did / or did not do to get it to work. Thanks Babak [EMAIL PROTECTED] wrote: Well I eventually got the 7940 loaded Now does anyone have quick fix to get it to work with asterisk Tar in advance Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, 9 October 2003 9:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem Had those same problems with some 7960's but not with others. As previously mentioned (below and by others on the list over the last year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x and somewhere around v4.x remove all the comment lines in SIPDefault. It will work. The problem has been observed by many. Since none of us have 7960 source code, we can only guess at what the root problem is. Or, you can put the phones under Cisco maintenance, call the TAC center, and they'll tell you to do the same thing. Thanks to all that have responsed on this issue. To clarify, I have been able to successfully upgrade at least 4 phones. 2 are giving me the malformed packet problem. They were both running MGCP 3.3 code by the way. The phone goes through it's standard boot sequence, then looks for the file OS79XX.TXT which it sees and downloads properly. This file only contains the image version I want the phoen to have: P0S3-05-3-00 It then tries to download the file but iserts some garbage characters in the name: P0S3-05-3-00Garbage.bin at which point it does not upgrade since the file name contains garbage characters. This is the second phone I have seen do this same exact thing. The first was a 7940, this is now a 7960. So in summary the process I have been using has been successful except with two phone which are doing the same exact thing. Thanks again for the attention to this matter. Babak Rich Adamson wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Without specific error codes, packet traces, etc, we can only guess. Some real options include: a) not all tftp servers are the same. Over about 15 years of experience, many tftp servers have an issue with the last packet and how to close the session. Might research the exact software you're using. b) the cisco phones seem to have an issue with the comment lines in the SIPDefault file (on the tftp server). Remove every one and test again. This seems to be highly dependent on the exact versions of SIP code implemented (in sequence) from v2.x upward. Some phones seem to have the problem while others don't. Since we don't know what version of code was in the phone when you received it, best guess is to boot to v2.x, then 3.x, then 4.x, then 5.x and I'll bet (at least a beer) the problem disappears. Why? don't know, but lots of similar comments on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by responding to [EMAIL PROTECTED] You are required to delete the contents and destroy any copies immediately. IGX Global is not liable for the views expressed in this electronic message or for the consequences of any computer viruses that may be unknowingly transmitted within this message. This electronic message is also subject to standard copyright/ownership laws. It is not intended to be reproduced, or re-transmitted without the consent of the originator. www.igxglobal.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ---End of
Re: [Asterisk-Users] pbx_spool and contexts
This had to do with a revision of request_and_dial, where the real bug lives. It's fixed in CVS now and the hack mentioned here should no longer need to be applied. Mark On Wed, 8 Oct 2003, Richard Lyman wrote: same issue as previously noted... look at lines 1628ish in chan_iax.c and line 1645ish in chan_iax2.c for strncpy(c-context, context, sizeof(c-context) - 1); or similar... comment those out with // disclaimer: not sure what else this breaks. Steve Creel wrote: When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop properly into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where pstn-number is a valid number) -- start -- Channel: IAX2/[EMAIL PROTECTED]/pstn-number MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: test-call Extension: s Priority: 1 -- end -- The problem surfaced after upgrading to current CVS (10/8) from 9/9. Is anyone else having this problem? Is there something I should be doing differently? Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sasquatch, the Loch Ness Monster, UFOs and...
TC wrote: Mythical Asterisk Creatures, oft-discussed, rarely seen: 1) An advanced graphical user interface 2) An IAX2 hardware device 3) A Radius CDR report module 4) A live-method, robust SQL-based dialplan 5) LDAP/SQL/Radius authentication for SIP phones 6) Robust R2 signalling support 7) Multilingual language recordings of all existing * .gsm files 8) Free exchange of PSTN gateways in a centralized routing arbiter model 9) Speech recognition support 10) Database abstraction module for ...CDR, SQL Dial Plan, DBGet/Put, * config files 11) SoftFaxModem 12) SS7 Signaling 13) WEB Interface for Users/Admin 6 is working here, though I have reasons not to release it just yet. 11 is basically working here. Expect something for general consumption before the month is out. The actual FAX modem part is now functional in both directions. I'm doing the protocol logic right now. I then need to integrate it with Asterisk. I have been transferring FAXs successfully between a real FAX machine and my software., though without Asterisk involved up to now (I am using an E400P, but I am not yet using Asterisk). I have implemented only the 1980 FAX features, where any patents ran out some time ago. If I can figure out whether there are any patent issues with newer features (e.g. 14,400 baud V.17), I might work on other unencumbered features later. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pbx_spool and contexts Still Not Working
On Thu, 9 Oct 2003, Mark Spencer wrote: This had to do with a revision of request_and_dial, where the real bug lives. It's fixed in CVS now and the hack mentioned here should no longer need to be applied. Mark On Wed, 8 Oct 2003, Richard Lyman wrote: same issue as previously noted... look at lines 1628ish in chan_iax.c and line 1645ish in chan_iax2.c for strncpy(c-context, context, sizeof(c-context) - 1); or similar... comment those out with // disclaimer: not sure what else this breaks. Steve Creel wrote: When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop properly into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where pstn-number is a valid number) -- start -- Channel: IAX2/[EMAIL PROTECTED]/pstn-number MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: test-call Extension: s Priority: 1 -- end -- The problem surfaced after upgrading to current CVS (10/8) from 9/9. Is anyone else having this problem? Is there something I should be doing differently? Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Using the latest CVS, (3 min ago) I am still getting these errors Oct 9 09:15:41 WARNING[1226054960]: File pbx.c, Line 1754 (ast_pbx_run): Channel 'SIP/mlh-2d67' sent into invalid extension 's' in context 'default', but no invalid handler Oct 9 09:15:41 NOTICE[1226054960]: File pbx_spool.c, Line 206 (attempt_thread): Call completed to SIP/mlh Below is my outgoing file: Channel: SIP/mlh MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: record Extension: s Priority: 1 As you can see, it is still not working with the context correctly. Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP softphone volume control?
Hi, Where are the settings to access the demo server at Digium? I would like to setup and test x-lite as well with a running asterisk until i get my box up and running. Thanks -- Original Message -- From: Chris Albertson [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 8 Oct 2003 10:53:12 -0700 (PDT) I went back to the example system direct from CVS with small additions to sip.conf and extnsion.conf needed to make one xten X-Lite phone work. I can dail in and hear the anouncements, call the demo server at Digium. The audio quality I hear comming from Asterisk back to X-Lite is good (9 on a 10 scale) but the sound volume comming from the X-Lite extension is very low even hard to hear. I know about the mic. level adjustment on X-Lite and I've got it set high almost to the point of clipping I appears that the Asterisk server is somehow scaling the sound down. Is this adjustable? Some way to set a per extension gain cotrol? = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 TFTP Problem
After about three hours I just TFTP to 7940 I had that weird file issue So renamed the file Using Cisco tftp unticked the box that says transfer this file only And after 50 or so attempts there you go Honestly I would rather load an IOS on our big mother routers ( if you know what I mean ) Now my issue is I can call in but can not get to the extension ( Cisco ) And from the Cisco phone I can not call out ( pstn ) Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Babak Pasdar Sent: Thursday, 9 October 2003 10:57 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 7940/60 TFTP Problem Mick, Can you please provide more detail on specifically what you did / or did not do to get it to work. Thanks Babak [EMAIL PROTECTED] wrote: Well I eventually got the 7940 loaded Now does anyone have quick fix to get it to work with asterisk Tar in advance Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Thursday, 9 October 2003 9:41 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7940/60 TFTP Problem Had those same problems with some 7960's but not with others. As previously mentioned (below and by others on the list over the last year), start by downloading sip v2.x, then v3.x, then v4.x then 5.x and somewhere around v4.x remove all the comment lines in SIPDefault. It will work. The problem has been observed by many. Since none of us have 7960 source code, we can only guess at what the root problem is. Or, you can put the phones under Cisco maintenance, call the TAC center, and they'll tell you to do the same thing. Thanks to all that have responsed on this issue. To clarify, I have been able to successfully upgrade at least 4 phones. 2 are giving me the malformed packet problem. They were both running MGCP 3.3 code by the way. The phone goes through it's standard boot sequence, then looks for the file OS79XX.TXT which it sees and downloads properly. This file only contains the image version I want the phoen to have: P0S3-05-3-00 It then tries to download the file but iserts some garbage characters in the name: P0S3-05-3-00Garbage.bin at which point it does not upgrade since the file name contains garbage characters. This is the second phone I have seen do this same exact thing. The first was a 7940, this is now a 7960. So in summary the process I have been using has been successful except with two phone which are doing the same exact thing. Thanks again for the attention to this matter. Babak Rich Adamson wrote: I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. I know why it does not work, but why are these malformed packets apearing. I also tried to go to a windows based TFTP server (original was linux TFTP) with the same results. Any ideas anyone? Without specific error codes, packet traces, etc, we can only guess. Some real options include: a) not all tftp servers are the same. Over about 15 years of experience, many tftp servers have an issue with the last packet and how to close the session. Might research the exact software you're using. b) the cisco phones seem to have an issue with the comment lines in the SIPDefault file (on the tftp server). Remove every one and test again. This seems to be highly dependent on the exact versions of SIP code implemented (in sequence) from v2.x upward. Some phones seem to have the problem while others don't. Since we don't know what version of code was in the phone when you received it, best guess is to boot to v2.x, then 3.x, then 4.x, then 5.x and I'll bet (at least a beer) the problem disappears. Why? don't know, but lots of similar comments on the list. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Babak Pasdar Founder/CTO IGX Global 389 Main St. Hackensack, NJ 07601 www.igxglobal.com (201) 498-0555 ext. 2205 The electronic message that you have received and any attachments are solely intended for the use of the addressee(s) and may contain information that is confidential. If you receive this email in error, please advise us by
Re: [Asterisk-Users] iax2 trunk
Im having problems setting up a trunk between two locations. Heres the
setup I have:
Server A is connected to the PSTN at my datacenter
Server B is connected to a clients e1 line at his datacenter
I only want to route calls from Server B to Server A and out through the
PSTN. Server A has a lot of other things connecting to it, so I need a
very specific context for all calls to go through. Because of the volume
of calls between the two servers I wish to setup a trunk.
Server A has this entry in iax.conf
[serverb]
type=friend
host=serverbipaddress
trunk=yes
auth=md5,plaintext,rsa
secret=s3rv3rb
username=serverb
context=serverb
qualify=yes
Server B doesnt have much in iax.conf - only codec and port information
under [general]
Server B is using this in his extensions.conf though:
exten = _X.,1,Dial,IAX2/serverb:[EMAIL PROTECTED]/${EXTEN}|180
now i know some things are wrong, i know i can use type=peer because its
only a one way connection (im not making calls to serverb, only recieving
calls from it)
but when i do an iax2 trunk debug i get this:
IAX2 Trunk Debug Requested
Beginning trunk processing
Processed trunk peer 'serverb' (0.0.0.0:0) with 0 call(s)
Ending trunk processing with 1 peers and 0 calls processed
even though there are calls going between the servers - so obviously they
arent using the trunking facility. so whats the deal. what do i have to
do in iax.conf on both sides and in extensions.conf on the side of ServerB
ok, bad form to reply to my own posting, but in case anyone else has this
problem in future it was a very silly setting. on server b i didnt need
the entry in iax.conf - all i needed was a register statement.
duncan
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[Asterisk-Users] concurrent calls
So whats the best way to find the maximum number of concurrent calls in this setup: IAX2 Trunk using GSM over a 512k internet line. thanks duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
show channels? On Thu, 9 Oct 2003, duncan wrote: So whats the best way to find the maximum number of concurrent calls in this setup: IAX2 Trunk using GSM over a 512k internet line. thanks duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
So whats the best way to find the maximum number of concurrent calls in this setup: IAX2 Trunk using GSM over a 512k internet line. show channels? actually i meant how to find out how many i could push down the 512k line - with regards to codec bandwidth and signalling etc... duncan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7940/60 TFTP Problem
-Original Message- From: Babak Pasdar [mailto:[EMAIL PROTECTED] I was wondering if anyone else has had this problem. I have purchused several Cisco 7940 and 7960 phones. Of the 5 phones so far I have run accross 2 that that give me malformed TFTP and refuse to upgrade to the latest version of SIP code -- 5.3. In fact some of the other phones also give malformed packets do this but they seem to work OK. The ones giving me the problem when looking in ethereal are misiing part of the filename to get it upgraded to SIP. Babak, Try a staged upgrade, especially is upgrading from a really old firmware. I.e., in OS79XX.TXT place an intermediate SIP image and then in SIPDefault.cnf put the image you wish to ultimately upgrade to. Here's my current files: [EMAIL PROTECTED] tftpboot]# more OS79XX.TXT P0S30202 [EMAIL PROTECTED] tftpboot]# more SIPDefault.cnf image_version:P0S3-04-4-00 proxy1_address: xx.xx.xx.xx tftp_cfg_dir: /configs/cisco7900/ proxy_register : 1 Phone starts off at Skinny 2.x, then upgrades to SIP 2.2 then to SIP 4.4. 4.4 could be assumably replaced by 5.3 HTH, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pbx_spool and contexts
hmm, i'd gone back thru ...request_and_dial to get to it... weird that i missed a simplier fix G Mark Spencer wrote: This had to do with a revision of request_and_dial, where the real bug lives. It's fixed in CVS now and the hack mentioned here should no longer need to be applied. Mark On Wed, 8 Oct 2003, Richard Lyman wrote: same issue as previously noted... look at lines 1628ish in chan_iax.c and line 1645ish in chan_iax2.c for strncpy(c-context, context, sizeof(c-context) - 1); or similar... comment those out with // disclaimer: not sure what else this breaks. Steve Creel wrote: When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop properly into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where pstn-number is a valid number) -- start -- Channel: IAX2/[EMAIL PROTECTED]/pstn-number MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: test-call Extension: s Priority: 1 -- end -- The problem surfaced after upgrading to current CVS (10/8) from 9/9. Is anyone else having this problem? Is there something I should be doing differently? Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Loading a TDM card!!
Hi all, I do have the same problem. Does this problem appear with the last versions of TDM's board (TDMx0B)? I have seen a bug in feedback state (see http://bugs.digium.com/bug_view_page.php?bug_id=087), but the description it's not equal. I work with: Linux:Debian woody Kernel2.4.20 gcc:2.95.4 Hardware: P-IV 2 GHz + 256 Mb Motherboard DFI AD77 Infinity Digium's Boards: X100P TDM10B Any idea ?? bye. - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, October 03, 2003 8:17 PM Subject: [Asterisk-Users] Help Loading a TDM card!! Is there anything special needed to load up a TDM10B card?? I got the card today.. Took it from the box, put it into a PCI slot.. connected the power to the card and booted the PC.. I have removed the X100P to avoid confusion and I have the following in the config files.. in /etc/zaptel.conf # For the X100P #fxsks=1 # For the TDM10B fxoks=1 in /etc/asterisk/zapata.conf [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=yes callprogress=yes callgroup=1 pickupgroup=1 relaxdtmf=yes ;X100P - FXO PCI card ;signalling = fxs_ks ;channel = 1 ;TDM10B FXS device context=local signalling = fxo_ks channel = 1 When I boot the PC it does NOT show an added card/device when kudzu ran.. No LED's light up.. When I try to load the driver with init script.. [EMAIL PROTECTED] root]# /etc/init.d/zaptel start Loading zaptel framework: [ OK ] Loading zaptel hardware modules: Running ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6) [FAILED] dmesg has this to say.. Zapata Telephony Interface Registered on major 196 Specify address with base=0xN Registered Tormenta2 PCI If I try with modprobe.. [EMAIL PROTECTED] root]# modprobe wcfxs /lib/modules/2.4.20-20.9/misc/wcfxs.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20-20.9/misc/wcfxs.o: insmod /lib/modules/2.4.20-20.9/misc/wcfxs.o failed /lib/modules/2.4.20-20.9/misc/wcfxs.o: insmod wcfxs failed dmesg says.. Zapata Telephony Interface Registered on major 196 Zapata Telephony Interface Unloaded Have updated to very latest CVS as well.. Anyone got any ideas?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] concurrent calls
try to read this: http://lists.digium.com/pipermail/asterisk-users/2003-June/014788.html senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
duncan wrote: actually i meant how to find out how many i could push down the 512k line - with regards to codec bandwidth and signalling etc... Measure the data rate on one call and divide 512k by it for a rough estimate.. If you want more accuracy make one call and measure the data rate, then two calls and then 3 calls and you should start to see a trend in the increse in traffic per call.. My tests on one call showed GSM to use about 34Kbps* so you should manage 14 concurrent calls.. As a suggestion look at using the iLBC codec which in my test on one call used 25Kbps* which will give you close to 20 calls in your 512K connection.. Later.. *Averaged and Rounded off ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Loading a TDM card!!
Rafael Gonzalez Lomeña wrote: Hi all, I do have the same problem. Does this problem appear with the last versions of TDM's board (TDMx0B)? I have seen a bug in feedback state (see http://bugs.digium.com/bug_view_page.php?bug_id=087), but the description it's not equal. I work with: Linux:Debian woody Kernel2.4.20 gcc:2.95.4 Hardware: P-IV 2 GHz + 256 Mb Motherboard DFI AD77 Infinity Digium's Boards: X100P TDM10B Any idea ?? bye. My issue is related to the motherboard.. I appears that the PCI2.2 spec has a 3.3v supply on the bus.. My motherboard in my dev Asterisk server is an older board so it seems that this 3.3v supply is not there.. When I put the card into my P4 it detects that there is something there I haven't tested any further than this bacasue Asterisk is not installed on my P4.. I guess I am going to have to buy a newer motherboard to go any further.,. If you find a solution let me know.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
On Thu, 9 Oct 2003, WipeOut wrote: actually i meant how to find out how many i could push down the 512k line - with regards to codec bandwidth and signalling etc... Measure the data rate on one call and divide 512k by it for a rough estimate.. If you want more accuracy make one call and measure the data rate, then two calls and then 3 calls and you should start to see a trend in the increse in traffic per call.. My tests on one call showed GSM to use about 34Kbps* so you should manage 14 concurrent calls.. As a suggestion look at using the iLBC codec which in my test on one call used 25Kbps* which will give you close to 20 calls in your 512K connection.. of course this is assuming that trunk=no (in IAX atleast) - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunking confirmation?
Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for trunking to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my trunking has not been working properly since I set it up (It does in fact work, I can make and recieve calls over the IAX connection without any apperent problems).. So a short while ago I opened up my conf files and tried setting the type to peer from friend (yes I reloaded *).. Guess what??.. The link no longer worked.. I then tried setting type to user and it is worked.. So my IAX link works as friend and user but not as peer.. Seeing as peer was specified as the requirement in order to get an IAX2 trunk to work properly I am a little confused.. Is my setup working when it shouldn't be or have I got something backwards?? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] concurrent calls
WipeOut wrote: duncan wrote: actually i meant how to find out how many i could push down the 512k line - with regards to codec bandwidth and signalling etc... Measure the data rate on one call and divide 512k by it for a rough estimate.. If you want more accuracy make one call and measure the data rate, then two calls and then 3 calls and you should start to see a trend in the increse in traffic per call.. My tests on one call showed GSM to use about 34Kbps* so you should manage 14 concurrent calls.. As a suggestion look at using the iLBC codec which in my test on one call used 25Kbps* which will give you close to 20 calls in your 512K connection.. Some pointers to whitepapers and online bandwidth calculators: http://www.voip-info.org/wiki-Bandwidth+consumption /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunking confirmation?
Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for trunking to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my trunking has not been working properly since I set it up (It does in fact work, I can make and recieve calls over the IAX connection without any apperent problems).. So a short while ago I opened up my conf files and tried setting the type to peer from friend (yes I reloaded *).. Guess what??.. The link no longer worked.. I then tried setting type to user and it is worked.. So my IAX link works as friend and user but not as peer.. Seeing as peer was specified as the requirement in order to get an IAX2 trunk to work properly I am a little confused.. Is my setup working when it shouldn't be or have I got something backwards?? Later.. Something isn't working right with your system, specifically. I have many IAX2 configurations set to type=friend and trunk=yes which work quite well. I will agree with Jeremy though and say that having large implementations with type=friend may cause you headaches in the future depending on what you want to offer to your customers. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunking confirmation?
Hi Jeremy,
The handbook says:
user: A user can place calls to or through the Asterisk server.
peer: A peer receives calls from the Asterisk server, but does not
place them
friend: A friend both sends and receives calls through the Asterisk
server. This makes the most sense for handsets or other station
devices. When in doubt use this type.
Something must be wrong here since the peer setting says that this type of
entity can only receive calls. So if we set both ends to peer then nobody
can make calls. Can you, or somebody else shed some light here please?
Thanks,
Ricardo
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, October 09, 2003 12:48 PM
Subject: Re: [Asterisk-Users] IAX2 Trunking confirmation?
From the chan_iax2 source (around line 3712):
if (!peer) {
ast_log(LOG_WARNING, Unable to accept trunked packet from '%s:%d':
No matching peer\n, intoa(sin.sin_addr), ntohs(sin.sin_port));
return 1;
}
A friend is both a user and peer. However, I would discurage the use of
a friend as it will severely restrict your dialplan, espcially once you
are dealing with more than just a couple Asterisk boxes.
Jeremy McNamara
WipeOut wrote:
Hi,
My question is in refernece to the posting by Jeremy McNamara here..
http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html
He states that in order for trunking to work the type has to be
peer.. When I set mine up I did so using type=friend just to make it
simple.. So when I read the above posting I thought well maybe my
trunking has not been working properly since I set it up (It does in
fact work, I can make and recieve calls over the IAX connection
without any apperent problems)..
So a short while ago I opened up my conf files and tried setting the
type to peer from friend (yes I reloaded *).. Guess what??.. The link
no longer worked.. I then tried setting type to user and it is
worked.. So my IAX link works as friend and user but not as peer..
Seeing as peer was specified as the requirement in order to get an
IAX2 trunk to work properly I am a little confused..
Is my setup working when it shouldn't be or have I got something
backwards??
Later..
___
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Re: [Asterisk-Users] IAX2 Trunking confirmation?
Jeremy McNamara wrote: A friend is both a user and peer. However, I would discurage the use of a friend as it will severely restrict your dialplan, espcially once you are dealing with more than just a couple Asterisk boxes. Jeremy, Can you elaborate on how using type=friend would restrict the dialplan.. Just so I am aware of the pitfalls.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunking confirmation?
John Todd wrote: Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for trunking to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my trunking has not been working properly since I set it up (It does in fact work, I can make and recieve calls over the IAX connection without any apperent problems).. So a short while ago I opened up my conf files and tried setting the type to peer from friend (yes I reloaded *).. Guess what??.. The link no longer worked.. I then tried setting type to user and it is worked.. So my IAX link works as friend and user but not as peer.. Seeing as peer was specified as the requirement in order to get an IAX2 trunk to work properly I am a little confused.. Is my setup working when it shouldn't be or have I got something backwards?? Later.. Something isn't working right with your system, specifically. I have many IAX2 configurations set to type=friend and trunk=yes which work quite well. I will agree with Jeremy though and say that having large implementations with type=friend may cause you headaches in the future depending on what you want to offer to your customers. JT John, My setup also works with type=friend and trunk=yes with no problems.. I was just wondering why the original email stated that type had to be set to peer for trunking to work... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunking confirmation?
TeleSIP wrote: Hi Jeremy, The handbook says: user: A user can place calls to or through the Asterisk server. peer: A peer receives calls from the Asterisk server, but does not place them friend: A friend both sends and receives calls through the Asterisk server. This makes the most sense for handsets or other station devices. When in doubt use this type. Something must be wrong here since the peer setting says that this type of entity can only receive calls. So if we set both ends to peer then nobody can make calls. Can you, or somebody else shed some light here please? Thanks, Ricardo This is also what I would like to know.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 Trunking confirmation?
tcpdump is the easiest way. From 1 call to 50 calls the number of packets should be about the same, and they should just get larger. Mark On Thu, 9 Oct 2003, Jared Smith wrote: On Thu, 2003-10-09 at 11:39, WipeOut wrote: [snip] He states that in order for trunking to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my trunking has not been working properly since I set it up (It does in fact work, I can make and recieve calls over the IAX connection without any apperent problems).. I think you may be confused as to what the trunking is. Just because you can make calls over IAX doesn't necessarily mean you have trunking working. (Trunking combines packets from multiple calls to reduce overhead.) Jared Smith ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7914
I am looking into the possibility of buying a Cisco 7960 with a 7914 expansion module. I know a lot of people are using the 7960, but I haven't read much about the 7914 and I was wondering if anybody has used this module with Asterisk? -- Thank you for your time __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914
I've been told that the SIP firmware cannot deal with the 7914, however I've never been able to try it for myself as the few 7914s I have laying around here have no interface cable and I am unable to find the pinout. Even TAC couldn't help me :( Jeremy McNamara jerk face wrote: I am looking into the possibility of buying a Cisco 7960 with a 7914 expansion module. I know a lot of people are using the 7960, but I haven't read much about the 7914 and I was wondering if anybody has used this module with Asterisk? -- Thank you for your time __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
WipeOut wrote: Jeremy, Can you elaborate on how using type=friend would restrict the dialplan.. Just so I am aware of the pitfalls.. :) Mark's words to me, when I was a newbie: [00:08] kram a user is to authenticate an incoming call [00:08] kram a peer is someone you send a call to [00:08] kram friend, of course, is both I'll give you a real world example that has happened to more than a few NuFone customers: NuFone Customer A orders a toll-free number and termination from us. Instead of following the example config we send he does: [NuFone] type=friend secret=his_secret host=switch-1.nufone.net. context=NANPA When NuFone sends the toll-free calls to his Asterisk box, they will land in HIS NANPA context, which is really confusing, but he does figure that fact out and is able to make both toll-free inbound and outbound calls work. Then a few weeks later he decides to pick up a regular DID from us. Now, his (above) configuration will fail for the regular DID inbound calls, but nothing else, because our regular DIDs do not come from switch-1.nufone.net. The proper way is to separate the tasks. Starting with the user: [NuFone] type=user secret=his_secret context=inbound This way he is not restricting the hostname/IP address where the user 'NuFone' can call in from. Plus, he now has a more logical context for all of his inbound calls. and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Yes, a friend is a very easy way to get things started and it can be made to work, but you will end up causing hair loss and/or heartburn trying to figure out why everything doesn't work the way you expect it to, when you go to add more complexity to your operation. The moral of the story is: Separate those tasks now, so you can avoid problems later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to disable native bridge of SIP-to-SIP calls?
canreinvite=no in the appropriate sip.conf user or peer. Jeremy McNamara Anton Tinchev wrote: I have incoming calls from cisco AS5350 that are placed in queue. Queue rings on agents with SIP phones, and native bridge cousing some problems(no call at all). When i go to queue from iax client everything is just fine. So how to disable native bridge of SIP to SIP calls? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
Jeremy McNamara wrote: WipeOut wrote: Jeremy, Can you elaborate on how using type=friend would restrict the dialplan.. Just so I am aware of the pitfalls.. :) Mark's words to me, when I was a newbie: [00:08] kram a user is to authenticate an incoming call [00:08] kram a peer is someone you send a call to [00:08] kram friend, of course, is both I'll give you a real world example that has happened to more than a few NuFone customers: NuFone Customer A orders a toll-free number and termination from us. Instead of following the example config we send he does: [NuFone] type=friend secret=his_secret host=switch-1.nufone.net. context=NANPA When NuFone sends the toll-free calls to his Asterisk box, they will land in HIS NANPA context, which is really confusing, but he does figure that fact out and is able to make both toll-free inbound and outbound calls work. Then a few weeks later he decides to pick up a regular DID from us. Now, his (above) configuration will fail for the regular DID inbound calls, but nothing else, because our regular DIDs do not come from switch-1.nufone.net. The proper way is to separate the tasks. Starting with the user: [NuFone] type=user secret=his_secret context=inbound This way he is not restricting the hostname/IP address where the user 'NuFone' can call in from. Plus, he now has a more logical context for all of his inbound calls. and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Yes, a friend is a very easy way to get things started and it can be made to work, but you will end up causing hair loss and/or heartburn trying to figure out why everything doesn't work the way you expect it to, when you go to add more complexity to your operation. The moral of the story is: Separate those tasks now, so you can avoid problems later. Jeremy McNamara Jeremy, That makes a lot of sence (although I will have to read it a few more times just to cement the concept).. I can already see where this would have become a problem for me in the not so distant future.. Thanks a lot.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Ringing from PSTN
Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Record App Paths
If I do something like exten = 1,1,Record(/someplace/somefile|gsm) It does not record I end up getting -- Executing Record(SIP/mlh-04d0, |gsm) in new stack exten = 1,1,Record(filename|gsm) it works great! Is there anyway that I can set the path in the record app...if not is there an easy change I (or someone else ) can make to source so that I can. Thanks for your help, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote: and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? What if you wanted that specific user to drop into a specific context so you could tailor what was accessable for that user. Simple idea would be like a account I had opened up on my system for a potential overseas employee. This person had no need to be able to make calls to anywhere our switch allowed. I dropped the user into a specific context that only allowed dialing of a few specific phone numbers. This is different than say anyone else in our organization who have full run of the switch. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Results SUSE 8.2 + server size
Hello All, Thanks to those that responded to my problem of compiling on SUSE 8.2. I was not able to get the compile done so decided to put RedHat 9 on this system. After getting a RedHat supported NIC and RedHat installed, Asterisk compiled cleanly, one SIP phone is connected and voice mail works. No other tests have been run yet. A couple of days ago, Michael Farnworth asked about the smallest system that was running Asterisk. This one is a Pentium 100, 32 MB RAM, 8 GB disk. I don't expect it to handle much load but for a test platform it seems ok to use while trying to find a low cost P4 system. Regards, Robert Friedrichshafen, Germany ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No Ringing from PSTN
You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
On Thu, 2003-10-09 at 15:20, Thorsten Lockert wrote: and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? What if you wanted that specific user to drop into a specific context so you could tailor what was accessable for that user. But that would be an *inbound* again -- the question was why specify a context for a *peer*... Not for a user or friend, where inbound is possible... Thorsten ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Ringing from PSTN
That does make a ringing sound, but any idea what's causing the problem? Stephen Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM compression tool
On Thu, 2003-10-09 at 15:51, George Lin wrote: Hi list, Can anyone suggest us what kind compression tool is best to compress a GSM file. And what kind compression ratio can be? This is a hard message to write with out unleashing the flame thrower. On this list it has been discussed many times that you can use sox or toast to convert to GSM. At least you should have issued a apropos gsm on the command line, or even a man -k gsm. That alone would have pointed you to toast. A little study of GSM information tells you that the codec produces 32.5 bytes of data per 20ms. So compression ratio depends on the format it was in to begin with. Generally speaking though, you should be working with 8k samples a second and therefore 20ms is 160 samples. You may even be using 8bit samples like everything else is. At this point 160 samples is 160 bytes that gets compressed to 32.5 bytes. On computer platforms, it is a pain to deal with half bytes. So on a unix system, it has been standardized that 32.5 will be null padded to 33 bytes even. On Crapdos, they decided that this is one of the few places they would try not to bloat. On Crapdos, they take 2 32.5 byte frames and bit shift the second down into the empty half left by the first frame and produce a 65 byte double frame. Oddly enough, the majority of this is all learned from reading the source code readily available already in the asterisk code base. It possibly could be even more easily been found by a simple google search. At the minimum, please go here and read for the next 10 or so minutes. http://kbs.cs.tu-berlin.de/~jutta/toast.html -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7914
According to cisco it's just a serial cable, can you just use a straight through cable? It looks like a standard phone-headset type cable, though shorter. Nick On Thu, Oct 09, 2003 at 12:57:50PM -0700, jerk face wrote: Well that sucks. What about using SCCP --- Jeremy McNamara [EMAIL PROTECTED] wrote: I've been told that the SIP firmware cannot deal with the 7914, however I've never been able to try it for myself as the few 7914s I have laying around here have no interface cable and I am unable to find the pinout. Even TAC couldn't help me :( Jeremy McNamara jerk face wrote: I am looking into the possibility of buying a Cisco 7960 with a 7914 expansion module. I know a lot of people are using the 7960, but I haven't read much about the 7914 and I was wondering if anybody has used this module with Asterisk? -- Thank you for your time __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and DMS100 Channelized T-1
We have a DMS100 that does not have PRI. So we're using a channelized T1 using WU-LAW, ESF and B8ZS coming from the DMS100 that's plugged into a Tormenta2 Quad T1 Card on my Asterisk Box running Debian 3.01(woody) with Kernel 2.4.22. The Link is up but according to the DMS100, Channel_1 goes into RMB (Remote Manual Block) and Channel_2 goes into LO (Lock Out). I've been through all my Asterisk Configs with help from BKW_ in #Asterisk on IRC and everything looks great there. The Tormenta2 Span 1 has a green light and asterisk loads each channel fine. I'm not sure where the problem is originating from so if anyone has any helpful insight or feedback they could provide on this matter it would be greatly appreciated. Some DMS100 working configs would be great too. Jason Helmich MIS, Blue Sky Communications PGP Key ID: 0x4CF71E92 [EMAIL PROTECTED] 011.684.258.1077 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record App Paths
Record(/tmp/testing:gsm) Thats what I use.. and it works. bkw On Thu, 9 Oct 2003, Lists wrote: If I do something like exten = 1,1,Record(/someplace/somefile|gsm) It does not record I end up getting -- Executing Record(SIP/mlh-04d0, |gsm) in new stack exten = 1,1,Record(filename|gsm) it works great! Is there anyway that I can set the path in the record app...if not is there an easy change I (or someone else ) can make to source so that I can. Thanks for your help, Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] my phone shows asterisk
What hardware are you using to connect to the PSTN? G At 07:35 AM 10/9/2003, listas iPfone [EMAIL PROTECTED] wrote: Hi all, When i receive a call from pstn ( calls from sip works well) my phone shows asterisk and not the number of the phone. How can i make asterisk show the phone number of the person who caled? thanks! Miklos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM compression tool
Thanks Steve. In fact, I am looking for a ZIP tool to zip a GSM file. currently I found that winzip ONLY compress 10% of a WAV file. I am wondering is there any good ZIP tool for a GSM file and or WAV file. Thanks, George Lin. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, October 09, 2003 2:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GSM compression tool On Thu, 2003-10-09 at 15:51, George Lin wrote: Hi list, Can anyone suggest us what kind compression tool is best to compress a GSM file. And what kind compression ratio can be? This is a hard message to write with out unleashing the flame thrower. On this list it has been discussed many times that you can use sox or toast to convert to GSM. At least you should have issued a apropos gsm on the command line, or even a man -k gsm. That alone would have pointed you to toast. A little study of GSM information tells you that the codec produces 32.5 bytes of data per 20ms. So compression ratio depends on the format it was in to begin with. Generally speaking though, you should be working with 8k samples a second and therefore 20ms is 160 samples. You may even be using 8bit samples like everything else is. At this point 160 samples is 160 bytes that gets compressed to 32.5 bytes. On computer platforms, it is a pain to deal with half bytes. So on a unix system, it has been standardized that 32.5 will be null padded to 33 bytes even. On Crapdos, they decided that this is one of the few places they would try not to bloat. On Crapdos, they take 2 32.5 byte frames and bit shift the second down into the empty half left by the first frame and produce a 65 byte double frame. Oddly enough, the majority of this is all learned from reading the source code readily available already in the asterisk code base. It possibly could be even more easily been found by a simple google search. At the minimum, please go here and read for the next 10 or so minutes. http://kbs.cs.tu-berlin.de/~jutta/toast.html -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Low Volume
When a call is placed connecting to the X100P, the volume of the call is very low. I have played with the gain settings without many results. Any suggestions? Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P setup in Switzerland
Hi all, I am trying to setup an E100P for use on Swisscom E1-PRI here in Switzerland. Swisscom seems to use Siemens hardware. Here are my configs (cvs from a few hours ago) : zaptel.conf loadzone=fr ; tried de but got warning at modprobe defaultzone=fr span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens SDSL-G703 modem switches its LEDs off ... not sure bchan=1-15,17-31 dchan=16 zapata.conf ... switchtype = euroisdn signalling = pri_cpe group = 1 channel = 1-15,17-31 When I try to dial-out, I get : NOTICE[33809]: File app_dial.c, Line 502 (dial_exec): Unable to create channel of type 'Zap' == Everyone is busy at this time -- Executing Congestion(SIP/22-8f32, ) in new stack == Spawn extension (from-sip, 32423423423, 2) exited non-zero on 'SIP/22-8f32' Any idea ?? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What is the pingtime option in iax chan(iax.conf)?
Sorry for asking for it, but it is nowhere documented. There is no maches in the mailing list or the whole google. I found it just in sources - conf parser of chan_iax.c. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk festival problem.
On Wed, 2003-10-08 at 18:12, Eric Wieling wrote: Does this work? festival_client --tts_mode Do you want to play a game? Yes. But since I dont have a soundcard in the box I use another tts command. I quote mi first email: Also I tested with festival_client executing the same command (tts_text blabla 'file) and I got a file in NIST format (8 Khz) which I converted to WAV and played it just fine. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk festival problem.
On Wed, 2003-10-08 at 18:12, Eric Wieling wrote: Does this work? festival_client --tts_mode Do you want to play a game? To be more specific I tried this command. festival_client --output jj.wav pp where pp is a file with the following command: (tts_textasterisk Hi there, how are you ? 'file)(quit) And I got a file pp.wav which was in NIST format so I converted it with sox to WAV and it heard fine. -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E100P setup in Switzerland
Quoting Marcel Prisi [EMAIL PROTECTED]: Swisscom seems to use Siemens hardware. Here are my configs (cvs from a few hours ago) : zaptel.conf loadzone=fr ; tried de but got warning at modprobe defaultzone=fr span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens bchan=1-15,17-31 dchan=16 First of all use ccs not cas, When starting asterisk do U see that D channel is up Are B channels restarted ? When running zttool do you have OK value on your E1 Can u see the status from siemens side ? regards m. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GSM compression tool
On Thu, 2003-10-09 at 18:26, George Lin wrote: Thanks Steve. In fact, I am looking for a ZIP tool to zip a GSM file. currently I found that winzip ONLY compress 10% of a WAV file. I am wondering is there any good ZIP tool for a GSM file and or WAV file. Don't expect to get much compression with lossless compression like zip or any other tools like that. GSM is a lossy compression and that is the way it gets some of the compression. GSM is probably the best you will get for compression and still be usable on your asterisk machine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Thursday, October 09, 2003 2:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] GSM compression tool On Thu, 2003-10-09 at 15:51, George Lin wrote: Hi list, Can anyone suggest us what kind compression tool is best to compress a GSM file. And what kind compression ratio can be? This is a hard message to write with out unleashing the flame thrower. On this list it has been discussed many times that you can use sox or toast to convert to GSM. At least you should have issued a apropos gsm on the command line, or even a man -k gsm. That alone would have pointed you to toast. A little study of GSM information tells you that the codec produces 32.5 bytes of data per 20ms. So compression ratio depends on the format it was in to begin with. Generally speaking though, you should be working with 8k samples a second and therefore 20ms is 160 samples. You may even be using 8bit samples like everything else is. At this point 160 samples is 160 bytes that gets compressed to 32.5 bytes. On computer platforms, it is a pain to deal with half bytes. So on a unix system, it has been standardized that 32.5 will be null padded to 33 bytes even. On Crapdos, they decided that this is one of the few places they would try not to bloat. On Crapdos, they take 2 32.5 byte frames and bit shift the second down into the empty half left by the first frame and produce a 65 byte double frame. Oddly enough, the majority of this is all learned from reading the source code readily available already in the asterisk code base. It possibly could be even more easily been found by a simple google search. At the minimum, please go here and read for the next 10 or so minutes. http://kbs.cs.tu-berlin.de/~jutta/toast.html -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Config
What is the proper method to install/configure an X100P FXO card?
Re: [Asterisk-Users] Cisco 7914
On Thu, Oct 09, 2003 at 03:28:21PM -0400, Jeremy McNamara wrote: I've been told that the SIP firmware cannot deal with the 7914, however This is correct, I just tried it, and there's no support for the 7914 expansion module in the SIP image. All I got is steady read light on the buttons. I've never been able to try it for myself as the few 7914s I have laying around here have no interface cable and I am unable to find the pinout. Even TAC couldn't help me :( I'll find out the pinout and post it tomorrow :) As a side note, the skinny image does support them, will it work with chan_skinny? Yifang ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] University phone system
I just talked with a friend that is a computer teacher at the local collage. He heard about my experiments with asterisk and some grandstream phones, and he wants to get a small setup going as a class project, which will hopefully expand to cover the whole building. Right now if a teacher needs to call, he needs to go to the teachers lounge and use the phone in there. Also, each room does not have a phone, which is getting very old (if the teacher needs to talk to another teacher, they have to actully walk to the other classroom). They also have many campuses scattered through out the state, and each has a direct T1 to each one, which would allow expansion to make campus to campus calls for free, without using up long distance lines. I was wondering if anyone has ever setup something like this, maybe with it starting out very small, (soft ip phones in the classroom to play around with *), and gradually growing into something larger. This would be good intro into linux, and would also benefit the campus. Any thoughts on this idea and tips would be appreciated. Doug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * consultant needed - will pay
Hi PJ- I specialize in large volume IVR systems both here and in Europe. (please see my web site Case Studies for more info) If it's just a simple IVR with database, I can likely do the demo very cheaply, to get a chance at the bigger job. Already have the AGI's and Perl script to accomplish this most likely. Please email me directly (off line), and we can talk when you like. Thanks, Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PJ Welsh Sent: Thursday, October 09, 2003 9:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] * consultant needed - will pay Thank you for reading this, sorry to waste bandwidth otherwise. I am part of a US company looking for someone to setup a demo IVR system for us. I seem unable with my current knowledge to pull this off myself. The demo is the regular enter your id and validate/repeat/continue methodoligy you put up with in everyday life. I would like to have the validation and other parts done via database (Postgres or MySQL). This is a FOR PAY job, with the potential for landing the full project. I need a quick turnaround! I have gotten myself in a serious time crunch before I have to go with another proposed M$ solution and a great deal more money. I need for contacts as soon as possible. I would ask that you be able to accept either PayPal or PO or work till you get a check. I DO NOT know how to handle the potential for transactions outside of the US. If you are outside the US and can still accept US $'s and know the implecations, I think we can work something out. I can provide additional info to interested people email to: pj at cassens*dot*com I will reply as soon as possible. Again, I need a quick turnaround! Skills in * + database + AGI are likely manditory. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Redhat system init and wcusb
Hi Tom- Someone may have already answered you on this, but if not: Didn't you receive a quick start sheet with your demo kit? It should cover this. If not, what works for me (also running Red Hat 9.0) is to add the following lines to the /etc/rc.d/rc.local file: rmmod usb-uhci modprobe usb-uhci modprobe wcfxo modprobe wcusb sleep 1 ztcfg -vv sleep 1 The above shell commands are executed by the system at the end of the re-boot process, so go ahead and re-boot... Then try starting asterisk by typing asterisk -c. Once you get asterisk starting ok this way, you could add the asterisk line to the rc.local file at the end to start everything on reboot. Hope this helps. Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tom Sent: Thursday, October 09, 2003 6:56 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Redhat system init and wcusb I have the dev kit lite installed, and after compiling and installing asterisk, I've become a little confused about how and when wcfxo and wcusb are loaded. When ever my box is done booting, the wcusb module is loaded, but the wcfxo module is not. Further, even though my startup notices say that asterisk started OK, it is not running by the time I log in and do a ps. First, where are the wcusb/wcfxo modules being loaded at boot time. It seems like this should happen in /etc/rc/rc.sysinit, but there is no reference to anything associated with asterisk. The asterisk init script doesn't load them either. Any help on this would be appreciated. Regards, Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
Thorsten Lockert wrote: and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Uh. Why would you want to specify a context for a peer at all...? Aren't those used only for inbound anyhow? No, you actually don't need to use a context in the peer. Asterisk will leave it up to the far end to decide what context to use. We use it to avoid any possibility of confusion in the process, but it is not necessary. In fact, I just verified this with the master himself and we will no longer tell our customers to use a context in their peer. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Ringing from PSTN
I set mine up like this exten = 1234,2,Dial(sip/[EMAIL PROTECTED],20,r) And everytime it rings I get exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 And * falls over This is with a voicetronix openline4 card Any ideas ??? Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Friday, 10 October 2003 6:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] No Ringing from PSTN You can send a fake ring by using something like: exten = 1234,1,Dial(SIP/[EMAIL PROTECTED],20,r) Assuming the ATA is in the sip.conf as [1234] However, this does NOT solve the underlying problem. On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote: Here is my Configuration PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with DTMF 'looping' / mis-dials (X100P card)
Hi all, I'm having a problem with * being very finicky about the length of DTMF key-presses during menus, voicemail, etc. Basically, short (100 ms) tones are ignored, anything between 100ms (or so) and about 300ms is correctly detected, and anything 300ms is interpreted as multiple presses of the same key. This is terrible for callers who are trying to get to the correct extension number, delete a voicemail message, etc. Any ideas why this is happening, or how to fix it? I searched the mailing list back to 7/1/03 but found no mention. Here's my * setup. Note presence of Vodavi Starplus DHS phone system in call path. Pentium II -350 / Redhat 9.0 / (3) X100P cards X100P cards are connected to an Analog SLT adaptor, which goes to a digital ports on a Vodavi StarPlus DHS phone system. (So call path is: PSTN -- Vodavi Starplus -- Analog SLT adaptor -- X100P card.) Asterisk CVS-08/29/03-09:23:49 I have not updated to the most recent CVS because of various problems I've seen cropping up on the bug tracking site.. Something tells me this is not CVS-related but perhaps something to do with the Vodavi. Any suggestions? DTMF parameters I can tweak? Here's an example of what Voicemail2 does when I hold down the 7 key while listening to a message: (Flip-flop period is about 3-4 cycles per second.) -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' -- Playing 'vm-deleted' -- Playing 'vm-undeleted' Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?)
- Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 09, 2003 2:57 PM Subject: Proper usage of user's and peer's (was Re: [Asterisk-Users] IAX2 Trunking confirmation?) WipeOut wrote: Jeremy, Can you elaborate on how using type=friend would restrict the dialplan.. Just so I am aware of the pitfalls.. :) Mark's words to me, when I was a newbie: [00:08] kram a user is to authenticate an incoming call [00:08] kram a peer is someone you send a call to [00:08] kram friend, of course, is both I am still at a loss here. If both are set to peer then how can either end originate the call? You would need at least one end to be user or friend. I'll give you a real world example that has happened to more than a few NuFone customers: NuFone Customer A orders a toll-free number and termination from us. Instead of following the example config we send he does: [NuFone] type=friend secret=his_secret host=switch-1.nufone.net. context=NANPA When NuFone sends the toll-free calls to his Asterisk box, they will land in HIS NANPA context, which is really confusing, but he does figure that fact out and is able to make both toll-free inbound and outbound calls work. Then a few weeks later he decides to pick up a regular DID from us. Now, his (above) configuration will fail for the regular DID inbound calls, but nothing else, because our regular DIDs do not come from switch-1.nufone.net. The proper way is to separate the tasks. Starting with the user: [NuFone] type=user secret=his_secret context=inbound This way he is not restricting the hostname/IP address where the user 'NuFone' can call in from. Plus, he now has a more logical context for all of his inbound calls. and the peer simply has the required information: [NuFone] type=peer secret=his_secret context=NANPA host=switch-1.nufone.net Yes, a friend is a very easy way to get things started and it can be made to work, but you will end up causing hair loss and/or heartburn trying to figure out why everything doesn't work the way you expect it to, when you go to add more complexity to your operation. The moral of the story is: Separate those tasks now, so you can avoid problems later. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
