RE: [Asterisk-Users] SMTP MTA suggestions.
I usually use Qmail www.qmail.org, in my humble opinion it is more straight forward to configure than sendmail. On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote: I understand asterisk invokes sendmail in order to send email notifications of messages left. Is there another application less complicated than Sendmail, I already got mail servers else where and they are the ones I want to use. Any light in this matter will be appreciated. There are several replacements, but sendmail isn't any harder to config. You usually only need to change 3 lines in the sendmail config. * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 'asterisk' displayed on my Cisco 7960 7912 ...
The problem is some calls from the PSTN have hidden caller id so if you want to change it to something else then modify chan_sip.c #define CALLERID_UNKNOWNAsterisk I've changed mine to: #define CALLERID_UNKNOWNUnknown -Original Message- From: Shaun Ewing [mailto:[EMAIL PROTECTED] Sent: 22 September 2004 14:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 'asterisk' displayed on my Cisco 7960 7912... On Wed, 22 Sep 2004 14:06:51 +0200, Evert Meulie [EMAIL PROTECTED] wrote: Hi! When I call a colleague of mine from my Cisco (via Asterisk), they get on their display: From Evert asterisk How do I remove/change the 'asterisk' part? Regards, Evert You need to set a valid caller ID number. For example, in sip.conf under the configuration for your phone: callerid=Shaun Ewing 7011 -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Remote-Party-ID
Marcello, This is something I am hoping for as well but I cannot find any features within the code to allow Asterisk to modify/create this field. Ideally I'd like to see the CallingPres function support Remote-Party-ID to disable/enable privacy. I actually placed a feature request some months back on bugs.digium.com but I don't think its been considered yet. Does anyone else have any experience with this that may help us ? Rgds, Adam -Original Message- From: Marcello Lupo [mailto:[EMAIL PROTECTED] Sent: 13 September 2004 11:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Remote-Party-ID Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide the number from the sip.conf for a particular user with restrictid=yes, i will get the call on the gateway from an anonymous caller and the switch will not get the callerid for accounting. If the * can put: Remote-Party-ID: number;party=calling;privacy=full;screen=yes The switch will interpret good this and will hide the number by himself. Any ideas? Thanks, Bye, MArcello * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call!
Ironic, Im just working on something similar myself, you can either use the appropriately named ex-girlfriend feature or I use GotoIf statements to match the caller id and maybe a timer or something to route to another context. ; note page search in girlfriend http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf ; note page search on CALLERIDNUM http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf ; found this too but havent used it http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetGroup -Original Message- From: Joseph Finley [mailto:[EMAIL PROTECTED] Sent: 13 September 2004 16:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Simple Caller-ID match and block and/or play voice file saying you're calling too much or don't call! The subject says it all. A couple of my sons have very annoying friends that tend to call ALOT. I usually don't like to answer the phone but these kids keep calling back with in 2 minutes of calling. I'm sure someone else has this problem and maybe using * to do a callerID match and block? Even add logic that if they called so many times in an hour? Or in my case, make it a month Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unknown RTP codec 72 received
According to IANA's list of RTP payload types (http://www.iana.org/assignments/rtp-parameters) RTP payload type 72 fulls within the following range: 72--76 reserved for RTCP conflict avoidance [RFC3550] I can't find much else in RFC3550 that defines it further but this should start you on the right path I hope. -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: 13 September 2004 16:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unknown RTP codec 72 received On Mon, 2004-09-13 at 06:13, Elman Efendiyev wrote: I get Unknown RTP codec 72 received message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 release d
Typo in your OS79XX.TXT P00 ? instead of P0S !? -Original Message- From: Michael Løjtnant [mailto:[EMAIL PROTECTED] Sent: 17 August 2004 13:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you can give me a hint. I added the files to my tftpd folder, changed the version-number in the file OS79XX.TXT - from P003-07-1-00 to P003-07-2-00 In my SIPDefault.cnf i changed the image_version from P0S3-07-1-00 to image_version: P0S3-07-2-00 Then I reboot it, and it loads the SIPMAC address.cnf - and reboots - and this goes on forever, or until i change the image_version number back to P0S3-07-1-00. What am i doing wrong - I just can't figure it out. Best regards Michael On Tue, 17 Aug 2004 16:28:52 +1000 Shaun Ewing [EMAIL PROTECTED] wrote: Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 release d
Ok please ignore me, I just tried 7.2 myself and worked fine with the same mods you made !? -Original Message- From: Michael Løjtnant [mailto:[EMAIL PROTECTED] Sent: 17 August 2004 13:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7.2 firmware for SIP 7940/7960 released Hi Shaun, Saw you post, and rushed to their ftp-server and downloaded it :-) But, I can't make my phone (7940) upgrade, so maybe you can give me a hint. I added the files to my tftpd folder, changed the version-number in the file OS79XX.TXT - from P003-07-1-00 to P003-07-2-00 In my SIPDefault.cnf i changed the image_version from P0S3-07-1-00 to image_version: P0S3-07-2-00 Then I reboot it, and it loads the SIPMAC address.cnf - and reboots - and this goes on forever, or until i change the image_version number back to P0S3-07-1-00. What am i doing wrong - I just can't figure it out. Best regards Michael On Tue, 17 Aug 2004 16:28:52 +1000 Shaun Ewing [EMAIL PROTECTED] wrote: Hi All, Just a heads up - I was looking around the Cisco FTP a little while ago and noticed that the SIP 7.2 images for Cisco IP Phone 7940/7960 were released yesterday (16th August). * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco PRI no CallerID
On Mon, 02 Aug 2004 20:23:24 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: * -- SIP -- CISCO -- PRI -- PSTN The PSTN sees no callerid. *--- PRI[zaptel]-- PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI-- * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri, pstn does see clid. help? A lot of carriers do CLI validation but it may also be as simple as the numbering plan/type that you are sending on outbound ISDN calls. Your carrier should of specified how they would like to receive the CLI (national/international format/preceeding zero maybe). As Jason said check with your carrier ... * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Routing incoming H.323 calls to specific contexts.
Hi, We've been working a lot with Asterisk in SIP for over 6 months but I've finally succumb to the pressure of H.323. I need to find a way to do what we do with SIP but with H.323. That is to have calls from H.323 peers placed into their own unique context (unique to the endpoint placing the call into Asterisk) within Asterisk so this is obviously done using REGISTER's within SIP but trying to do this with H.323 seems more challenging. I've installed GNUGK and have successfully had a H.323 device authenticate with the GNUGK and place calls onwards to Asterisk but I am unable to figure out how to place those calls into a unique context per H.323 endpoint/device/account without using their CLI to do so. I'm sure the community have solved this issue before, any help would be much appreciated and example configs would be perfect. Thank you in advance, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
I had a similar issue when installing my G.729 licences. I contacted Digium support and an engineer logged into my system and performed some hocus pocus and got it working for me ... -Original Message- From: Derek Samford [mailto:[EMAIL PROTECTED] Sent: 25 March 2004 18:29 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G.729 and SCSI If memory servers, and everyone feel free to flame away if it serves badly, the library only searches hda,hdb,hdc, and hdd. Try switching where your controller is, that may solve it. Derek -Original Message- From: Sergio Serrano [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 12:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G.729 and SCSI Yes I have mounted CDROM first with automount(/dev/cdrom) and second manually(/dev/hde) but nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andrew Thompson Enviado el: jueves, 25 de marzo de 2004 17:59 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk AGI - Redirect not sufficient, need to link channels
Thanks will give that a try also trying to patch the Link function into an AGI command as well ... -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: 22 March 2004 18:49 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Asterisk AGI - Redirect not sufficient, need to link channels You could simply redirect both of them to a meetme room with the 'q' flag set for no messages. I'm using that method for an application right now. MATT--- -Original Message- From: Low, Adam [mailto:[EMAIL PROTECTED] Sent: Monday, March 22, 2004 12:15 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Asterisk AGI - Redirect not sufficient, need to link channels Hey All, I'm developing a reception style console (like many others) to answer incoming calls to a main line number, request who they want to speak to and then have the receptionist call the desired party and announce the calling party before putting them through. This should be fairly straight forward except for the fact that I end of with two channels an no way to bind them together. I've search the source code long and hard and am unable to find a way to hack something quickly together. I am sure others have hit this issue, does anyone have any advise ? Rgds, Adam _ Adam J. Low Tel: +31 20 778 2740 Senior Network ArchitectFax: +31 20 778 2600 Priority Telecom Corporate Email: [EMAIL PROTECTED] * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk AGI - Redirect not sufficient, need to link channels
Hey All, I'm developing a reception style console (like many others) to answer incoming calls to a main line number, request who they want to speak to and then have the receptionist call the desired party and announce the calling party before putting them through. This should be fairly straight forward except for the fact that I end of with two channels an no way to bind them together. I've search the source code long and hard and am unable to find a way to hack something quickly together. I am sure others have hit this issue, does anyone have any advise ? Rgds, Adam _ Adam J. Low Tel: +31 20 778 2740 Senior Network ArchitectFax: +31 20 778 2600 Priority Telecom Corporate Email: [EMAIL PROTECTED] * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VXML_URL and Cisco 7960 Phones?
I tried to get that working as well and also found it was not available in the SIP image. You can't do pushes either to the phone like you can with SCCP. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: 14 March 2004 13:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] VXML_URL and Cisco 7960 Phones? I was tempted by the wiki that mentions the (very undocumented) VXML_URL and suggests it might be able to control the display on a Cisco phone during an incoming call using a SIP image. I've mucked around with this for over two hours and after scouring source code, google, and the archives have found nothing. Does anyone have any how to use this feature? Does it even really exist? I can see the header being set and hitting the phone - but I can't find documentation anywhere suggesting what format you can send it. It's my understanding, although I've no direct experience, the function does not exist in the SIP images. The limitation is highly likely related to Cisco marketing plans and not to real design/programming capability, etc. (How else would one sell proprietary systems?) Anyone have a disassembler? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice star tsafter ring.
Has anyone reported a bug for this ? if so what's the id ? -Original Message- From: Andrew Thompson [mailto:[EMAIL PROTECTED] Sent: 11 March 2004 23:02 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 and short delay before voice startsafter ring. Steve Dolloff wrote: We have the same complaint here. The caller doesn't hear the receiver say hello and so no-one knows what's going on. Stephen I get this also, on my Sipura SPA-2000. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Codecs [G.729]
I posted the results of my real world analysis of codec bandwidth usage on this list a couple of weeks back. Here's the table I put together and an example of calculating bandwidth over ADSL. G.711 over Ethernet = 95 Kbps per channel G.711 over IP/PPP = 86 Kbps per channel G.711 over ADSL/ATM = 108 Kbps per channel G.729 over Ethernet = 39 Kbps per channel G.729 over IP/PPP = 30 Kbps per channel G.729 over ADSL/ATM = 45 Kbps per channel 200 bytes G.711 packet + 8 bytes AAL5 overhead = 208 bytes 208 bytes fit in 5 cells of 48 bytes payload 5 cells are 265 bytes. VoIP over ATM AAL5MUX thus has an overhead of 21.51% VoIP G.711 conversation sends 50 packets per second. This uses 250 cells per second. This causes approximately 10 OAM5 cells to be sent over the duration. The total bitrate is thus (250 + 10) * 53 bytes * 8 bits = 110240 bits/second = 107.66Kbit/s -Original Message- From: Rich Adamson To: [EMAIL PROTECTED] Sent: 10-3-04 12:41 Subject: Re: [Asterisk-Users] Asterisk Codecs [G.729] All of the numbers he's showing are apparently adding inbound and outbound traffic together, giving results that are approximately double what is actually seen on the wire. If he is working in a half-duplex ethernet environment, those numbers have some meaning; if full-duplex, then cut them in half for reasonable engineering values. (Also, some _appear_ to be questionable.) What is the method you are using to test the bandwidth. Can you give us a outline how to do a bit rate measurement on asterisk. snip ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Have anyone test it with G.729? Please let me know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
Well I just took a look at the TAC case and things dont look good, seems the TAC are now blaming Asterisk for the problem but I will go through there debugs and push back, will let you know. -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 08 March 2004 22:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring. Thanks for the information. You have saved me a few hours on the phone with TAC. smile Low, Adam wrote: We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 and short delay before voice star ts after ring.
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated) but now we are hearing that it will not be fixed in that release but would most likely be further down the track. The issue is specific to SIP on 79xx phones, H.323/SCCP/MGCP all work fine. We're hoping to get a *special* release of the bug fixed SIP code for testing within the next 3/4 weeks. If we get it I'll post an update ... -Original Message- From: Duane [mailto:[EMAIL PROTECTED] Sent: 03 March 2004 15:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7960 and short delay before voice starts after ring. Bisker, Scott (7805) wrote: I think what James is referring to is the delay once the call already been dialed. It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. The problem is that once a call is dialed, when the remote party picks up the phone, the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. I noticed the same issue using a SIP soft phone, I can't recall having the same issue with a IAX soft phone, pretty sure it didn't happen... I'm testing now to see if I can make it happen, but it seems to be fine... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] calls being presented as Anonymous
Reece, I have a similar setup by the sounds of things (running 0.7.2 with AS5300) and on private number calls what you actually get is 'Anonymous 010101010101' and as far as I remember it was always like that for me. How are you pulling the callerid into your script ? -Original Message- From: Reece Anderson To: [EMAIL PROTECTED] Sent: 3/4/04 5:41 PM Subject: [Asterisk-Users] calls being presented as Anonymous Hi, Recently I upgraded Asterisk from version 5 to 7 since I've done this all the calls that are private numbers are now showing up as Anonymous. I know for a fact its not the Cisco 5300 striping this off it appears to be Asterisk itself. Does anyone know the section of source code that needs modifying to re-enable this, we currently have an identification system on a few 100 numbers through a database, this currently is not matching to any clients. I'd prefer to not downgrade as instability issues forced the upgrade in the first place. Private posts welcome if this information is not suitable for the mailing list :) * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office requirements - Can this be done ?
I've done a fare amount of analysis on codec bandwidth requirements and you should remember that you typically will require more bandwidth over ADSL than you would over any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711 channel rather than 86kb over straight PPP/HDLC based connections. Why I hear you ask ? The following calculations are based on G.711 PCM running at 20ms samples resulting in 200 byte packets (default for most codec implementations). 200 bytes G.711 packet + 8 bytes AAL5 overhead = 208 bytes 208 bytes fit in 5 cells of 48 bytes payload 5 cells are 265 bytes. VoIP over ATM AAL5MUX thus has an overhead of 21.51% VoIP G.711 conversation sends 50 packets per second. This uses 250 cells per second. This causes approximately 10 OAM5 cells to be sent over the duration. The total bitrate is thus (250 + 10) * 53 bytes * 8 bits = 110240 bits/second = 107.66Kbit/s Steve Kennedy wrote: On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from Covad (in the US Southwest) and I have sustained 4 calls without a problem. I prefer to use GSM over G.711to squeeze it down, but that is my choice. I don't feel that call quality is substandard. That's the crunch (1.5/512) ... it's actually the 512 which is relevent. Virtually all DSL in the UK is a wholesale product from BT (they have about 2 million customers, Easynet who local loop unbundle may have 20,000, the rest of the providers maybe another 10,000 between them). All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 and 50:1, but actually a lot less than that), there are a few providers doing their own contention over BT's product. However the 256K upstream is still the limiting factor, so you can get one, and MAYBE two VoIP lines over it. If BT would up the upstream to 512, you could probaly get 4 out of it Steve On the UK DSL using G.711 you should easily get 2 concurrect calls, G.711 uses about 84k(incl overhead) in each direction, so 2 calls would be 168K (of the 256k) If you switch o GSM or iLBC you should get 6 concurrent calls, and if you were to use IAX2 trunking you could *maybe* squeeze another one.. Other codecs could offer even more but I haven't tested them.. Later.. * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Small office requirements - Can this be done ?
So for us Dummies out here :) who just know it works. Yep, it sure does, I thought it was something people might find interesting. Its certainly been a challenging subject for me to try and provide reliable and high quality voice service over ADSL. In my experience it seems to depend a hell of a lot on the QoS deployed on the ATM network behind the DSLAM's. Obviously a single cell being dropped every 5 cells would effectively cause every G.711 IP packet to be lost. Here in Holland I ported my KPN (legacy incumbent) telephone number to my home VoIP service about 4 months ago. It has been running over a BBNED ADSL service and works great 99.9% of the time. Although during recently virus/worm outbreaks I have found people complain they hear my voice choppy, probably due to the contention of all the other ADSL connection upstreams as they propagate those viruses/worms. This would mean that if you had a 512/256 aDSL and a 256 ISDN connection you would be able to have more channels over the ISDN? Thats right, I am not aware of any ADSL providers that actually provide their stated service level at an IP layer rather than at the ATM layer but maybe they are out there ... The exact calculation depends on how your encapsulating IP over the 256k ISDN connection. I will assume your actually getting 4x B channels with either multi-link PPP (haven't calculated the overhead for this one) or a CSU/DSU converting to X.21/V.35 (preferable). You should be able to push 3 concurrent G.711 channels over that 256k ISDN service assuming 86Kbps per channel. David Here's a little table I put together for our capacity planning team: G.711 over Ethernet = 95 Kbps per channel G.711 over IP/PPP = 86 Kbps per channel G.711 over ADSL/ATM = 108 Kbps per channel G.729 over Ethernet = 39 Kbps per channel G.729 over IP/PPP = 30 Kbps per channel G.729 over ADSL/ATM = 45 Kbps per channel * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS login
I have the same issue ... -Original Message-From: Glenn Dalgliesh [mailto:[EMAIL PROTECTED]Sent: 01 March 2004 16:03To: [EMAIL PROTECTED]Subject: [Asterisk-Users] CVS login I seem to be having trouble with cvs login. anyone having similar problems It just hangs after entering the password * 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
RE: [Asterisk-Users] CVS login
Or perhaps I should say 'adams.psknet.com' is down, box appears to be down ... -Original Message- From: Low, Adam Sent: 01 March 2004 16:14 To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] CVS login It seems to me that 63.171.251.202 (adams.psknet.com) is problematic and 65.38.23.22 (ns.bkw.org) is ok ... [EMAIL PROTECTED]:/usr/src]$ cvs login Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot CVS password: 11:32:06.909911 am00devel01.33537 adams.psknet.com.cvspserver: S 2588535351:2588535351(0) win 5840 mss 1460,sackOK,timestamp 847565593 0,nop,wscale 0 (DF) 11:32:09.901669 am00devel01.33537 adams.psknet.com.cvspserver: S 2588535351:2588535351(0) win 5840 mss 1460,sackOK,timestamp 847565893 0,nop,wscale 0 (DF) -Original Message- From: John Fraizer [mailto:[EMAIL PROTECTED] Sent: 01 March 2004 16:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CVS login Glenn Dalgliesh wrote: I seem to be having trouble with cvs login. anyone having similar problems It just hangs after entering the password Make sure you actually have connectivity to the CVS server (ping/traceroute). John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecificallyCLID priva cy
Stephen, Thanks for the suggestion but my problem is with inbound calls from the PSTN (coming in via a AS5300) into the SIP based platform and how the * chan_sip identifies that a PSTN originated call should have the number withheld or not. Rgds, Adam -Original Message- From: Steve Dolloff [mailto:[EMAIL PROTECTED] Sent: 26 February 2004 22:12 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy I have the following in my sip.conf entries: callerid=Anonymous 8885551212 This still passes the number for 911, but flags the call as private. I believe this will meet your requirements. Stephen -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specificallyCLID priva cy Low, Adam wrote: Hey All, I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones. My issue is that from what I see in chan_sip.c there is no support for the Remote-Party-ID field in relation to withholding the calling partys number. This is a legal requirement for many countries and although it doesnt appear as an Impressed. Does some countries have laws on SIP implementations? Wow. ;-) Is this something planned to be added or perhaps a minor oversight ? If it's somethine planned to be added is really up to your (our someone else's) willingness to code... :-) Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=full Could you please point me in direction of standard documents, drafts or documentation of this? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
Impressed. Does some countries have laws on SIP implementations? Wow. ;-) We operate a large traditional telephone network in several countries and as I am sure you are aware lawful intercept is a requirement on traditional networks. We've extended our network to provide VoIP gateways (SIP/H323 based) into our traditional Nortel based switched network and even though the calls may originate from a SIP/H323 based network that does not remove the legal requirement within the traditional switched network to abide by the rules of our telecoms licence. The law maybe immature in relation to regulation of SIP/H323 voice networks but those wishing to interconnect with traditional voice switched networks will still have to abide by the applicable rules/laws if they wish to send traffic over the PSTN. Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy
Well I am in mostly a Cisco enviroment and it seems that it is supported on both IOS 12.3(4)T for the AS5300 and the SIP6.2 image on our 7940's. I've not tested any other SIP stacks but maybe others can offer some added input there ? Ok I'll submit it to bugs.digium now ... -Original Message- From: Olle E. Johansson [mailto:[EMAIL PROTECTED] Sent: 27 February 2004 12:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, sp ecifically CLID priva cy Low, Adam wrote: Could you please point me in direction of standard documents, drafts or documentation of this? IETF specification, draft-ietf-privacy-.02.txt, SIP Extensions for Caller Identity and Privacy. Thank you for the pointer, as this is still a draft (a lot of SIP things are), it's not a requirement to implement it. And it may be too early to do so, since drafts may change. Do you know any more products supporting this? I'll download the draft and look into it. Please open a request on http://bugs.digium.com so we don't loose it in the large amount of traffic on the list. Having it in bugs keeps it in place and we could continue the discussion in there. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best VOIP Analog adapter ???
I've been testing a nice little box that has precisely what you requested. Its made by Aethra (Spain) I believe and know as the VIP3001 or VIP3002 and it runs both SIP/H323 and allows you to select if you want to send calls of the VoIP or over the PSTN. It works great with Asterisk running SIP. Although I just tried to find it on their website and its not there so I think it might be that I have a beta testing unit. Adam -Original Message- From: Carlos Arnt [mailto:[EMAIL PROTECTED] Sent: 27 February 2004 15:15 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best VOIP Analog adapter ??? Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog phone . Anyone know some adapter that make this miracle ? Thanks alot, Carlos ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip support for SIP:Remote-Party-ID, specifically CLID priva cy
Hey All, I have a Cisco AS5300 running SIP against an Asterisk server with multiple C7940 phones. My issue is that from what I see in chan_sip.c there is no support for the Remote-Party-ID field in relation to withholding the calling partys number. This is a legal requirement for many countries and although it doesnt appear as an issue on the actual C7940 handsets when the Voicemail email is sent out it does contain the calling partys supposedly 'hidden' calling party id. Is this something planned to be added or perhaps a minor oversight ? Rgds, Adam Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=full _ Adam J. Low Tel: +31 20 778 2740 Senior Network ArchitectFax: +31 20 778 2600 Priority Telecom Corporate Email: [EMAIL PROTECTED] * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cannot find -lXext when building * ?
As Tilghman indicated X is definitely not required to build Asterisk, we run RH9 without any X related packages installed and it compiles and runs perfectly. -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: 17 February 2004 19:17 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] cannot find -lXext when building * ? Tilghman Lesher wrote: You have GTK installed, but not X? If you don't intend to run X applications on the server, then deinstall GTK (as the X libraries are required to run GTK apps). This leads to a question that has been bugging us for a while. Is X required to build asterisk? When we tried to do so on machines that didn't have any X libraries installed, we would get errors at link time building pbx.c If we removed the lines that called for those libraries, asterisk would build, but then other weirdnesses would ensue and we finally just started installing the X libraries even though we will never run X on those boxes (low-end servers talking to one or two channels). Could anyone comment on the right way to do it? It's got to be either me or the code :-) Thx. b. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Analog Cordless Phone Recommendations
I don't think there is really an issue with 'which' analog phone, the only issue (I am aware of) with interoperability is in relation to CallerID. In the US it seems FSCK (I understand from my Aussie colleague that FSCK is also used in Australia) is always used and across Europe it seems to be DTMF but the actual format of the DTMF varies from country to country. Features such as stuttered dial tone are generated by the FXS interface. Like Dan I use ATA186's which generates the stuttered dialtone when messages are waiting and its completely separate from the handset. I have no experience with the TDM10B but I am confident it can do all that the ATA can do ... FYI: I am lucky enough to be using the BO BeoCom 2 which works great with the ATA. Rgds, Adam -Original Message- From: Christopher Lee [mailto:[EMAIL PROTECTED] Sent: 17 February 2004 09:00 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Analog Cordless Phone Recommendations Hi all, I've just added a TDM10B (1port FXS) to my Asterisk box and want to use this extension with a cordless phone. In particular I'm just wondering if anyone has any suggestions for a phone that will perhaps be able to detect voicemail waiting on the Asterisk server? I'm guessing I should be able to get asterisk to generate a stuttered dial tone when a message is waiting, so it's just a matter of finding such a cordless phone that can detect this. Cheers, Chris Lee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 - how to enable messages key
Hmmm did you read any of the docs on cisco.com ? You need to set the 'message_uri' option to the extension that you run VoiceMailMain on into the configuration file (SIP000XXX.cnf) for the phone. -Original Message- From: John Fraizer To: [EMAIL PROTECTED] Sent: 11-2-04 6:22 Subject: Re: [Asterisk-Users] Cisco 7960 - how to enable messages key Um, tell it what to do? I don't remember exactly what I did but, it was intuitive enough that when I got my 7960 a week ago, it only took one try to get it right. Paul Mahler wrote: Does anyone know how to make the 7960 messages key dial voicemail? SIP 6.0. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vegastream 50 FXO with Asterisk
I have a Vega 50 BRI working without any of the issues you mentioned, the dual SIP registrations is normal for most multi-line boxes enabled split users. Rgds, Adam -Original Message-From: Glenn Dalgliesh [mailto:[EMAIL PROTECTED]Sent: 05 February 2004 20:11To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Vegastream 50 FXO with Asterisk Anyone have any experienceconfiguringVegaStream's with Asterisk. Ihave run into afew of questions. 1. It appear that after turning on registrations I am seeing two request for registration per linesip:[EMAIL PROTECTED]sip:[EMAIL PROTECTED]What is purpose and how do I handle this?2. DTMF btw Asterisk and the Unit I was unable to get rfc2833 to work successfully with inbound or outbound DTMF. Is this a known issue? 3. How is the best way to deal with dialout and selecting a free channel on the VegaStream Any general suggestions/experiences with regard to configuring a VegaStream withasteriskwould be appricated.Thanks * 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] Asterisk + oh323 docs ?
Does anyone have any documentation on Asterisk + oh323, I am trying to allow a H323 peer to send me calls that I want to push out to SIP phones but am having trouble passing the digits dialed from the oh323 peer and dialing those digits onto a SIP client. Any docs much appreciated or even better working extensions.conf Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Record conversation
res_monitor.so: Resource for recording channels. -Original Message-From: Rattana BIV [mailto:[EMAIL PROTECTED]Sent: 05 February 2004 16:20To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Record conversation Hi, Does anybody know if it is possible to record a conversation with asterisk ? Regards Rattana * 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
RE: [Asterisk-Users] P2P RTP without SIP re-invites
Several people have requested more information on my cluster setup, I'll try to put something together today but things are very busy here at the moment ... but keep an eye for a mail today ... -Original Message- From: David Luyens [mailto:[EMAIL PROTECTED] Sent: 03 February 2004 07:39 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] P2P RTP without SIP re-invites Hi Adam, could you share your clustering setup? David * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] P2P RTP without SIP re-invites
Apologies for the belated reply but I've spent the weekend fighting DDoS attacks against Superbowl sites ... )c; Ok, well I am not sure what went wrong with previous testing but I have tried this again with Cisco 7940's and Cisco AS5300's and indeed the RTP stream flows directly between end-points retaining SIP signalling via Asterisk. This is exactly the operation I had hoped for. I had previously tested with my home 7940 which it behind NAT without success and so will re-test this this evening. Thanks for all the responses and related discussion on clustering Asterisk, thanks to those I now have a running cluster of 3 Asterisk servers each with mirrored sip.conf and extensions.conf built dynamically from a MySQL backend database. Rgds, Adam -Original Message- From: Brancaleoni Matteo [mailto:[EMAIL PROTECTED] Sent: 31 January 2004 13:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] P2P RTP without SIP re-invites hi I guess this would work if both Alice and Bob were NAT'ed on the inside of the same NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes and they're on separate NAT'ed networks, the call is broken. So it's a dangerous configuration. nope. I have a public * server (beta server for a free VoIP service), on a public IP. and some sip phones around , like one in my home, behind nat, one in my office (another nat) and some others at my coworkers home... all behind nat. and are different nat box, do you agree? that works ok, I have RTP passing directly from one endpoint to the other... no RTP on the public * server. No stun is used. The phones are budgetones in this case. All are configured with nat=yes on asterisk side. or I missing something? -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] P2P RTP without SIP re-invites
I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. So with that assumption I imagine a platform that would not get involved with the actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each end of the call deal with RTP encoding with their dedicated DSP hardware. There is an alternative in mind that maybe I could utilise some old Dialogic DSP cards that we have but I suspect trying to get these working with Asterisk would be a lot of programming work that I probably couldn't manage, maybe I'm wrong ? The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd (specifically SIP breaks and calls are not torn down correctly) and of course you lose a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when it is in the SIP signalling path. I vaguely remember previous discussions on this and even a patch but I am unable to find anything in the archives, does anybody have any info on that ? The conclusion I have come to is that I would try and patch the Asterisk code. The idea being that when the RTP parameters are negotiated that Asterisk would pass through the source address/port from each SIP client causing them to talk RTP directly. I intend to begin work on this this weekend but am I hoping that maybe somebody else has already achieved what I desire, anybody ? Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Need Europian vendor for Digium hardware.
http://www.digium.com/index.php?menu=resellers#Europe -Original Message- From: Anton Tinchev [mailto:[EMAIL PROTECTED] Sent: 26 January 2004 11:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Need Europian vendor for Digium hardware. Must accepts wire transfers and ships to Sofia. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Remote reload Cisco 7960
You need a little more to make this script reboot the phone. It basically instructs the phone to check a file called 'syncinfo.xml' at its TFTP URL. This file needs to contain the following line: IMAGE VERSION=* SYNC=2/ The number 2 above is the sync value which must be different (I think higher) than the sync: field defined in your SIPDefault.cnf file. Then the script should do its stuff and reboot the phone. Rgds, Adam -Original Message- From: B. J. Bomar [mailto:[EMAIL PROTECTED] Sent: Monday, January 19, 2004 6:42 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Remote reload Cisco 7960 I've tried to use that script, but the phones seem to ignore it. I am in the process of upgrading to 6.1 on the phones, maybe they will behave like they're supposed to. B. J. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Friday, January 16, 2004 22:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Remote reload Cisco 7960 http://www.bkw.org/~brian/cisco/reboot7960.txt or you can us this handy perl script.. NEXT!!! bkw On Fri, 16 Jan 2004, Rich Adamson wrote: Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. telnet to the box and reload it. command line has the ability. rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re-Invite between SIP phones
canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re-Invite between SIP phones
I'd suggest placing a packet sniffer (tcpdump, etherreal) and see whats happening because it works great for me and always has but I guess it also requires support on the end-points and possibly (assuming non-cisco enviro) there maybe an option that needs to be configured on your phones/gateways. Please provide more information on your setup ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:52 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re-Invite between SIP phones Already did that, but it's not working. Al --- Low, Adam [EMAIL PROTECTED] wrote: canreinvite=yes within sip.conf entities ... -Original Message- From: Al [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 20, 2004 2:06 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re-Invite between SIP phones Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATA186 SIP Outbound Fax Calls
All, I was wondering if anyone has any experience with the Cisco ATA186 (SIP image) and outbound faxing with Asterisk. Inbound faxs from PSTN into * and on to the ATA work fine but outbound faxs receive congestion from *. I've got packet dumps from both sides and everything appears normal but after about 3 seconds the * servers sends the AS5300 a CANCEL and sends the ATA a '503 Service Unavailable' (with CSeq: 2 INVITE). The ATA responds with a SIP 2 ACK but does not stop sending RTP packets but the * server has taken down its RTP state so responds with ICMP port unreachables. I've disable all fax tone detection on the ATA and AS5300 but still can't seem to get this to work. If anyone has any advise or recommended ATA configs it would be much appreciated. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris
Florian, Sorry you haven't heard anything but we've recently decided not to offer this product out side of Holland. If your still interested we have another product called ISDN-Flex that provides SIP/H.323 PSTN access inbound/outbound but you need to be connected on on one of our IP or MetroLan products so we can guarantee the QoS. Rgds, Adam -Original Message- From: Florian Overkamp [mailto:[EMAIL PROTECTED] Sent: Monday, December 01, 2003 7:00 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris At 16:46 1-12-2003 +0100, you wrote: Well the Aussie's recently announced an additional travel warning for The Netherlands due to the increased level of petty crime although I feel it was a little extreme. The petty crime problem is very much specific to Amsterdam and foreign crims come into the city specifically to target tourists and their valuables. I've lived out here for 3 years now and enjoy exceptional safety where I live in Haarlem so perhaps an alternative major city such as Haarlem or Den Haag might be an option ? How about Enschede ? ;-) BTW Adam, kick your people please, I still haven't heard anything from them :-P Best regards, Florian ___ 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
[Asterisk-Users] Dedicated * voicemail server
Hey All, I've started to try and distribute the functionality of my single * server amongst a few varying servers. The issue I have is that when splitting out the voicemail portion onto a dedicated server I am no longer able to inform the voicemail application (when call originated from a different box) if the call hitting the voicemail server was sent there because it was unanswered or if the phone was busy. I'm not sure if there is something within IAX that can pass this information on from one * server to another or if there is another solution ? Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dedicated * voicemail server
You could add an initial digit based on whether it was a busy or no answer forward, use the extra digit to determine the message played on the VM server and just strip it back off to get the mailbox number. Email me direct if that isn't clear enough. This is actually what I have at the moment, the prepend was an issue because I've tried to make the platform non-national specific so every mailbox has its full telephone number including country code. Instead I am prepending but this makes things dirty and complicated. I was hoping for some magical SIP option or something but I guess I'll proceed with the current setup. * 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] Re: Asterisk European Tour: was RE: * Party in Paris
Second that ! -Original Message- From: Cees de Groot [mailto:[EMAIL PROTECTED] Sent: Monday, December 01, 2003 2:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk European Tour: was RE: * Party in Paris zoa [EMAIL PROTECTED] said: And while you are in Europe, why not also do Brussels ? ;) Amsterdam!! -- Cees de Groot http://www.tric.nl [EMAIL PROTECTED] tric, the new way helpdesk/ticketing software, VoIP/CTI, web applications, custom development ___ 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] Re: Asterisk European Tour: was RE: * Party in Paris
Amsterdam!! I had my laptop and suitcase stolen in Amsterdam the one time I went there, after hearing someone talk about how safe a city it was over dinner. Most importantly, also stolen was my (apparently irreplacable) copyleft shirt (yellow/gold with large blue backwards (C) symbol on front and GPL preamble on back) which no amount of effort has managed to find a replacement for and it's *that* part i've never really gotten over. Mark Well the Aussie's recently announced an additional travel warning for The Netherlands due to the increased level of petty crime although I feel it was a little extreme. The petty crime problem is very much specific to Amsterdam and foreign crims come into the city specifically to target tourists and their valuables. I've lived out here for 3 years now and enjoy exceptional safety where I live in Haarlem so perhaps an alternative major city such as Haarlem or Den Haag might be an option ? Hmmm... what size was that T shirt ? (c; * 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: [offtopic] Re: [Asterisk-Users] Re: Asterisk European Tour: w as RE: * Party in Paris
Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot Actually we just had dinner and had left our things in his car which (according to the police inspector) was entered through the trunk using a half a tennis ball. Mark Yep I have seen it done, its amazing, place half a tennis ball over the lock (with specific central locking systems from almost all manufacturers) and give it a punch and the air pressure does its magic ... * 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] Re: Asterisk European Tour: was RE: * Party in Paris
Those things generally happen in Amsterdam. And in Kristiania in Copenhagen. The usual problem: Smoking too much pot I have to object to that, as a rule of thumb the Dutch only rob tourists who are dressed like tourists and act like tourists, that's what we all agreed to here and live by -- please just dress local and act local, so we can finally stop smoking pot just to keep up foreign misconception ... :-) Regards, Hans Vledder The Netherlands Hans although your somewhat right I don't think its fare to ask all tourists to leave their clothes at customs and to don clogs and ride a battered old bike around the city. I also must say that from my experience its very rarely (I've never heard of it) the native Dutch that perform these crimes. Sorry for the off topic ... * 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] Network Voip Carrier Termination (Off Topic)
We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Rgds, Adam -Original Message- From: reseaux [mailto:[EMAIL PROTECTED] Sent: 13 November 2003 13:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Network Voip Carrier Termination (Off Topic) Hi to ALL my name is Dimitri and im a CEO of startup Company in Italy focused on Internation call traffic i usualy use Asterisk (very good app :-) ) for switching call. I ask now to Asterisk User of Telecom Company if is possible to cooperate in creation of network International POP call Termination through Voip Tunnel from us. What we think about? Thanks to all Dimitri Bellini ___ 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] Network Voip Carrier Termination (Off Topic)
Low, Adam wrote: We can offer SIP based VoIP call termination in The Netherlands, Austria and Norway. If you'd like to speak to an account representative please contact me personally by email. Hmmm, this information should be on a website somewhere... Your probably right and it soon will do, because of my work with Asterisk (and general VoIP tech) our company has agreed that as we are the second largest PSTN provider in The Netherlands (we also operate in Norway and Austria) we should leverage our large switched telephone network by providing SIP/H.323 access to it. It's a brand spanking new product, the product team are still trying to get their heads around it but as soon as they do there will be a press announcement and documents posted on our site. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Feature request {with begging} sip debug ip_address
Hi *ers, If anyone with the capability and more appropriately the time, fancies developing a patch to provide sip debug ip_address capability with Asterisk I am sure they will be eternally praised (c; Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Encrypting SIP Phones
Hi Bryan, I am aware that the IETF have an Internet Draft in the pipelines for SRTP which can provide encryption and there is a lib out there available at: http://srtp.sourceforge.net/srtp.html I guess the real question would be if there is any intension to include this (or an equivelant) in the Asterisk source tree. I personally hope there is ... Rgds, Adam -Original Message- From: Bryan Nolen To: [EMAIL PROTECTED] Sent: 22/10/03 09:16 Subject: [Asterisk-Users] Encrypting SIP Phones Has anyone ever heard of such a beast? do they exist? (soft or hard phone) I am referring to the encrypting of the RTP data as the SIP headers will need to be read by asterisk still Bryan Nolen Lead Developer http://Arc.Net.AU http://arc.net.au/ http://cdonline.com.au http://cdonline.com.au/ * 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] Quick summary of Grandstream survey results
I'm not sure if it would be really practical to have a built in switch (although useful) within the phones. You really don't want your phone worrying too much about switching other ethernet frames whilst a call is in progress, you will probably then run in to queueing problems as you need to ensure your voice frames get priority. Even with the Cisco 79xx phones, you get significantly degraded performance on the PC side 100bT interface, most likely due to the lack of switching power. We found in our office enviroment that users soon began complaining of slower network connections and so we ended up reverting back to dedicated switch ports for the phones. Adam -Original Message- From: Andrew Kohlsmith To: [EMAIL PROTECTED] Sent: 22/10/03 04:18 Subject: Re: [Asterisk-Users] Quick summary of Grandstream survey results 100MB-ports 4 The list looks great but I just want to make mention of this one specifically -- Putting an El-Crappo 100mbit switch in there isn't going to do any better than what the BudgetTone 102 does now -- if they are going to do this, please please please encourage them to put a QUALITY switch in there (one that can sustain 100mbps) -- no use in even trying if it's going to degrade the network connection to the computer and make me install dual wiring anyway... Thank you for taking the time to post the list and gather up responses... These are decent phones and could go a LONG way with a few adjustments. Regards, Andrew ___ 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] Survey: Grandstream improvements.........
I don't have a single client that runs 10Mbps ethernet in their offices anymore and to tell them that the phone will downgrade their network speed to 10Mbps puts them off the phone straight away.. Hey WipeOut, Maybe I am missing something here but why would it downgrade their network speed to 10mbps, its very rare to find a 100bT switches these days that don't also support 10bT. In a switched ethernet network there would be no performance loss for the other ports !? * 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] Channel Banks
Well I disagree, there are numerous companies providing E1 channel banks, my personal favourite is J-tech of which I can find the damn link to their page for now ... Digging ... A quick google with e1 channel banks also found: http://www.valiantcom.com/vcl_cb/vcl_cb.html -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 10 October 2003 15:27 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channel Banks Mark Evans wrote: Hi All Can you's give me your thoughts on the best channel banks to use? Which are the easist to setup and which are the most reliable. Thanks Mark You may know already but the vast majority of channel banks are T1 only and typically only available in the US.. At least this is what I found when I was looking at using one.. Of course the dual mode card from Digium removes alot of the problem now in that you can have one or two ports of T1 to a CB and another port E1 to the PSTN.. ___ 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] 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
[Asterisk-Users] RE: Asterisk list a SPAMer (uol.com.br), I think not ...
Title: Message All, seems I too am suffering from posts to the list and being accused of SPAMing -Original Message-From: AntiSpam UOL [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 20:48To: [EMAIL PROTECTED]Subject: RE:RE: [Asterisk-Users] RTP routing.. Olá,Você enviou uma mensagem para [EMAIL PROTECTED]Para que sua mensagem seja encaminhada, por favor, clique aqui Esta confirmação é necessária porque [EMAIL PROTECTED] usa o Antispam UOL, um programa que elimina mensagens enviadas por robôs, como pornografia, propaganda e correntes.As próximas mensagens enviadas para [EMAIL PROTECTED] não precisarão ser confirmadas*.*Caso você receba outro pedido de confirmação, por favor, peça para [EMAIL PROTECTED] incluí-lo em sua lista de autorizados. Atenção! Se você não conseguir clicar no atalho acima, acesse este endereço:http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Hi,You´ve just sent a message to [EMAIL PROTECTED]In order to confirm the sent message, please click here This confirmation is necessary because [EMAIL PROTECTED] uses Antispam UOL, a service that avoids unwanted messages like advertising, pornography, viruses, and spams.Other messages sent to [EMAIL PROTECTED] won't need to be confirmed*.*If you receive another confirmation request, please ask [EMAIL PROTECTED] to include you in his/her authorized e-mail list. Warning! If the link doesn´t work, please copy the address below and paste it on your browser:http://tira-teima.as.uol.com.br/challengeSender.html?data=""> Use o AntiSpam UOL e proteja sua caixa postal * 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
RE: [Asterisk-Users] RE: Asterisk list a SPAMer (uol.com.br), I t hink not ...
Thanks, annoying but only course of action I guess ... (c; -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 29 September 2003 10:36 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Asterisk list a SPAMer (uol.com.br), I think not ... Just add a filter to your mail client to delete all mail from AntiSpam UOL [EMAIL PROTECTED].. Worked for me.. Low, Adam wrote: All, seems I too am suffering from posts to the list and being accused of SPAMing -Original Message- *From:* AntiSpam UOL [mailto:[EMAIL PROTECTED] *Sent:* 26 September 2003 20:48 *To:* [EMAIL PROTECTED] *Subject:* RE:RE: [Asterisk-Users] RTP routing.. http://antispam.uol.com.brhttp://www.uol.com.br Olá, Você enviou uma mensagem para [EMAIL PROTECTED] Para que sua mensagem seja encaminhada, por favor, *clique aqui* http://tira-teima.as.uol.com.br/challengeSender.html?data=ETG T78UpKvCZhfzyhxiozNotrKuM3O7574cCqPkZtVHDZSDMVAYPNODGLek2lC%2B 6T8FyvhBUIZsA%0ApU2rXIyJDH2tO3eNaWIJ%2BSThmoun81Dsx8vRmvKoeoAd YPeL8YIgrKgwQr1oRP0xrXAgzXvgIw%3D%3D Esta confirmação é necessária porque [EMAIL PROTECTED] usa o Antispam UOL, um programa que elimina mensagens enviadas por robôs, como pornografia, propaganda e correntes. *As próximas mensagens enviadas para [EMAIL PROTECTED] não precisarão ser confirmadas*.* *Caso você receba outro pedido de confirmação, por favor, peça para [EMAIL PROTECTED] incluí-lo em sua lista de autorizados. *Atenção!* Se você não conseguir clicar no atalho acima, acesse este endereço: http://tira-teima.as.uol.com.br/challengeSender.html?data=ETGT 78UpKvCZhfzyhxiozNotrKuM3O7574cCqPkZtVHDZSDMVAYPNODGLek2lC%2B6 T8FyvhBUIZsA%0ApU2rXIyJDH2tO3eNaWIJ%2BSThmoun81Dsx8vRmvKoeoAdY PeL8YIgrKgwQr1oRP0xrXAgzXvgIw%3D%3D -- -- Hi, You´ve just sent a message to [EMAIL PROTECTED] In order to confirm the sent message, please *click here* http://tira-teima.as.uol.com.br/challengeSender.html?data=ETG T78UpKvCZhfzyhxiozNotrKuM3O7574cCqPkZtVHDZSDMVAYPNODGLek2lC%2B 6T8FyvhBUIZsA%0ApU2rXIyJDH2tO3eNaWIJ%2BSThmoun81Dsx8vRmvKoeoAd YPeL8YIgrKgwQr1oRP0xrXAgzXvgIw%3D%3D This confirmation is necessary because [EMAIL PROTECTED] uses Antispam UOL, a service that avoids unwanted messages like advertising, pornography, viruses, and spams. *Other messages sent to [EMAIL PROTECTED] won't need to be confirmed*.* *If you receive another confirmation request, please ask [EMAIL PROTECTED] to include you in his/her authorized e-mail list. *Warning!* If the link doesn´t work, please copy the address below and paste it on your browser: http://tira-teima.as.uol.com.br/challengeSender.html?data=ETGT 78UpKvCZhfzyhxiozNotrKuM3O7574cCqPkZtVHDZSDMVAYPNODGLek2lC%2B6T8FyvhBUIZsA%0ApU2rXIyJDH2tO3eNaWIJ%2BSThmoun81Dsx8vRmvKoeoAdYPeL8YIgrKgwQr1oRP0xrXAgzXvgIw%3D%3D Use o *AntiSpam UOL* http://antispam.uol.com.br e proteja sua caixa postal ** 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 * 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] cisco AS5300 : problem configuration
I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly for me. -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: 29 September 2003 14:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco AS5300 : problem configuration Hi all !!! I m trying to setup a cisco AS5300 and I ve got some problem !!! During a call test I m getting this error message all the time. NOTICE[15371]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = kiki; Default for incoming calls allow=alaw ; Allow codecs in order of preference ;allow=ilbc ;allow=all [gw] type=user host=213.232.xxx.xx dtmfmode=rfc2833; Choices are inband, rfc2833, or info context=kiki -- Also when I allow all for the codecs that's doesn't work and in the SIP trace, it seems that Asterisk doesn't choose the appropriated codec. WHY ??? I really see the GW asking to use ulaw !!! -- When I try to setup a AGI script, for example: SAY DIGITS 7565 I can hear the first number 7 but nothing else !?! Any ideas about those problems ??? Thx for your helps, Areski ___ 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] cisco AS5300 : problem configuration
Areski, I would suggest you change the password on that 5300 right now, you provided the whole config file with the IP of AS5300 and the VTY password (although in very easy to break MD5) !!! Also in your sip.conf you have 'bindaddr = 0.0.0.0' so unless your running multiple NIC's on that box I'd suggest you comment out the bindaddr line altogether. -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: 29 September 2003 17:08 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration Hello, Below the IOS config file. Should I disable RFC3389 ??? If yes HOW ?? Show running-config - version 12.2 service timestamps debug datetime msec service timestamps log datetime msec service password-encryption service internal ! hostname UK-GW01 ! enable secret 5 $1$Q7QI$wgMvyRdFRxalCmgcEv7A81 ! ! ! resource-pool disable ! ip subnet-zero no ip domain lookup ! ! isdn switch-type primary-net5 ! voice call carrier capacity active ! ! ! ! ! ! ! ! ! mta receive maximum-recipients 0 ! controller E1 0 clock source free-running pri-group timeslots 1-31 ! controller E1 1 clock source line secondary 1 pri-group timeslots 1-31 ! controller E1 2 clock source line secondary 2 pri-group timeslots 1-31 ! controller E1 3 clock source line secondary 3 pri-group timeslots 1-31 ! ! ! interface Ethernet0 no ip address shutdown ! interface Serial0 no ip address shutdown no fair-queue clockrate 2015232 ! interface Serial1 no ip address shutdown no fair-queue clockrate 2015232 ! interface Serial2 no ip address shutdown no fair-queue clockrate 2015232 ! interface Serial3 no ip address shutdown no fair-queue clockrate 2015232 ! interface Serial0:15 no ip address ip mroute-cache isdn switch-type primary-net5 isdn incoming-voice modem no cdp enable ! interface Serial1:15 no ip address ip mroute-cache isdn switch-type primary-net5 isdn incoming-voice modem no cdp enable ! interface Serial2:15 no ip address ip mroute-cache isdn switch-type primary-net5 isdn incoming-voice modem no cdp enable ! interface Serial3:15 no ip address ip mroute-cache isdn switch-type primary-net5 isdn incoming-voice modem no cdp enable ! interface FastEthernet0 ip address 213.232.105.12 255.255.255.0 duplex auto speed auto ! ip classless ip route 0.0.0.0 0.0.0.0 213.232.105.254 no ip http server ! ! ! snmp-server community public RO snmp-server enable traps tty ! call rsvp-sync ! voice-port 0:D ! voice-port 1:D ! voice-port 2:D ! voice-port 3:D ! ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 100 pots application session direct-inward-dial port 0:D ! dial-peer voice 101 pots application session direct-inward-dial port 1:D ! dial-peer voice 102 pots application session direct-inward-dial port 2:D ! dial-peer voice 103 pots application session direct-inward-dial port 3:D ! dial-peer voice 300 voip application session destination-pattern 1879 progress_ind setup enable 3 session protocol sipv2 session target ipv4:62.39.85.18:5060 dtmf-relay rtp-nte codec g711alaw bytes 80 ! dial-peer voice 201 voip application session destination-pattern 1[6,7,9].. progress_ind setup enable 3 session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711alaw bytes 80 ! dial-peer voice 204 voip application session destination-pattern 18[0-6,8,9]. progress_ind setup enable 3 session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711alaw bytes 80 ! dial-peer voice 206 voip application session destination-pattern 187[0-8] progress_ind setup enable 3 session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711alaw bytes 80 ! gateway timer receive-rtcp 1000 ! sip-ua no oli sip-server ipv4:62.39.85.19:5060 ! ! line con 0 line aux 0 line vty 0 4 password 7 094D4210160B login ! end On Mon, 2003-09-29 at 14:17, Low, Adam wrote: I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly for me. -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: 29 September 2003 14:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco AS5300 : problem configuration Hi all !!! I m trying to setup a cisco AS5300 and I ve got some problem !!! During a call test I m getting this error message all the time. NOTICE[15371]: File rtp.c
RE: [Asterisk-Users] RTP routing..
WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ 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] RTP routing..
Hi, I work for an ISP (c; So I am going to build over the weekend a single Asterisk (RH9) box with two IP addresses (separate subnets) on the same NIC with a L2 ethernet switch connected then two separate routers (one for each subnet) and then try and make some calls to my production Asterisk box. I'll run EtheReal on the same L2 switch so I can see all the SIP and RTP packets so once its setup it should only take a few test calls to figure out exactly whats going on ... Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 13:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP routing.. Hi Adam, No queuing won't be an option.. all the traffic I am thinking about will be voice traffic moving in and out of the Asterisk box.. Are you setting up this same senario where you are boing to have two data paths?? Later.. Low, Adam wrote: WipeOut, Well will you really run out of bandwidth ? Would that be due to other (normal Internet traffic) traffic or would it all be RTP traffic, I ask because maybe some kind of priority queuing might be more effective ... It's a good question, the source and destination address/port of RTP packets is negotiated with SIP and I strongly suspect that Asterisk will only ever provide the primary address of an interface as the source (although this maybe be adjustable with bindaddr config option). I've just built a new Asterisk box so am going to try this out myself ... Will let you know ... Rgds, Adam -Original Message- From: WipeOut [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 11:36 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RTP routing.. Here is a question for all you routing guru's out there.. I am using an ADSL line (512/256Kbps) to connect from the internet to my Asterisk server.. At a point I will run out of bandwidth so the cheapest option would be to add a second ADSL line.. The problem is how will the routing work? If I put 2 IP's on one NIC will the return traffice be routed back via the gatway of the IP that is was recieved on or will it try and route all outbound traffic via the primary IP's gateway?? Would it be better to add 2 NICs instead of 2 IP's on one NIC?? although I don't see that this would change the routing logic.. Has anyone played with this type of setup? later.. ___ 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] RTP routing..
WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RTP routing..
I can restrict the RTP ports used with my Cisco 79xx phones and on my Cisco AS5300 and I think you can with Asterisk by using the rtp.conf but I'm not completely sure, I'd suggest diving into the source for that one ... -Original Message- From: Andre Lomonaco [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 14:31 To: '[EMAIL PROTECTED]' Subject: RES: [Asterisk-Users] RTP routing.. Hi, Sorry for my bad english but I´ll try to explain my problem I got an Asterisk running in my house with ADSL... I´m using X100P and TDM400P cards My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here´s the map with Firewalls Call for anyone to my house = PSTN = X100P = EXTENSIONS = SIP/RTP = ISA MICROSOFT FIREWALL = COMPUTER IN MY WORK WITH XLITE It´s working very nice, but I had to disable iptables in my Asterisk Box(Home)... I was using my linux with PPPoe Client, DynamicDnsClient and IPTABLES... I´d like to know if is possible to using IPTABLES again. My stupid question is: Can I restrict the ports that Asterisk uses to transmit RTP. When I was using IPTABLES with only port 5060 open , the SIP registration works nice but I didn´t receive sound... Andre Lomonaco -Mensagem original- De: Low, Adam [mailto:[EMAIL PROTECTED] Enviada em: Friday, September 26, 2003 9:06 AM Para: '[EMAIL PROTECTED]' Assunto: RE: [Asterisk-Users] RTP routing.. WipeOut, I just started to whiteboard this and had some realisations/questions: 1. I guess/hope your ADSL connection is not NAT'd ? 2. You will need two NIC's as I assume you will have two separate next hop gateways with each ADSL connection! 3. How would you load balance the inbound calls over the two connections (ensuring each doesn't exceed capacity)? The more I think about this the more I feel that a better solution would be to place a router between the Asterisk server and the two ADSL modems with some kind of NAT setup ... Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] FCC/Euro/Aussie approvals on TE410P
Excellent news, congratulations !! -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED] Sent: 26 September 2003 15:38 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FCC/Euro/Aussie approvals on TE410P I just got back from Boston where we completed testing of the TE410P for FCC, Euro, and Australian approvals, and I'm happy to say we passed all our approvals (including Q.921 and Q.931 layers, i.e. libpri as well as surges) for both telco and leased line applications. Hopefully we'll have the official documents soon, but I know there are a lot of you out there that are happy to hear that. Mark p.s. We were the *first* independent PRI implementation to come through that lab! Of all the units they've tested, we're the first to choose the build path on build vs. buy. ___ 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] Start of all recordings cut off
Slow machine? H I think its time I invested in hardware but my PII works great ! -Original Message- From: Peter Pauly [mailto:[EMAIL PROTECTED] Sent: 12 September 2003 12:29 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Start of all recordings cut off On Thu, Sep 11, 2003 at 09:30:35PM -0700, John Todd wrote: Before running any application that has sound playback (Playback, Background, VoiceMailMain2, etc.) it would be wise to execute an Answer first, then a Wait(2) to allow for VoIP channels to fully establish and settle. Adding Answer had no effect. Adding Wait(1) solved the problem. Maybe it's because Asterisk runs on a slow machine (750Mhz P3). ___ 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] Legal Interception - tapping
My 5 cents ... Since the ideal situation would be real-time monitoring then maybe a more effective solution would be to sample/duplicate the packets in the IP layer rather than expecting Asterisk to perform yet another auxiliary function. Cisco like most vendors are in a position were they have to provide Lawful Intercept capabilities within their own (VoIP IP) platforms very quickly to support the new European regulations. As a result of this a new feature will soon be available in Cisco IOS allowing routers (or AS5300's for that matter) to copy all inbound/outbound packets onto another interface or even re-write the destination address providing the capability to 'sniff' all IP (RTP/SIP) packets and route them off to another box. That other box could be another instance of Asterisk dedicated for the purpose or purely a replicated real-time packet stream routed directly to the authorities intercept platforms. -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent: 12 September 2003 04:33 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Legal Interception - tapping -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, September 11, 2003 10:20 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Legal Interception - tapping pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) issue. If they are using Asterisk is it not possible to record calls automatically. I have not reviews the CALEA requirements, must access be Yes it is very possible to record calls with *. I record all in and outbound calls. bkw I phrased that incorrectly, I have way too much email to look at I know it is possible to record calls, it will record them to a directory you define on the server. But are you required to provide archives/recordings of the calls or permit real-time tapping? ___ 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] Voicemail menu structure
This looks good to me, much better than the ilogical Cisco Call Manager voicemail menu structure ... -Original Message- From: Don Pobanz [mailto:[EMAIL PROTECTED] Sent: 12 September 2003 15:21 To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Voicemail menu structure There has been discussions about the voicemail menus and some of us would like to see an overall plan for the voicemail menus. There are 3 primary ways of arranging the menus. First is a tree structure, second is a random access structure and the third would be a hybrid of the two. (Comedian mail is currently a hybrid.) As was pointed out by Brad Bergman, the ideal would be to have it configurable in voicemail.conf as to whether to use the tree or the random or a hybrid structure. My assumption is that it would not be practical to make every key in the tree or every code for the random configurable. So, focusing only on the tree structure, what should the menus look like? Attached is a rough draft of what it may look like. Don Pobanz * 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] Cisco 7940/7960 XML application hint
I've been building a number of applications (SMS gateway, 411 directory interfaces, blah blah) recently along the same lines, I am mostly using Perl/MySQL and of course using the Cisco XML interface. I noticed people requesting more information on the XML interface and so I thought I'd drop a note for those interested. Most of the XML information is available on Cisco's site but there is also a Perl module specifically designed for people creating applications for the 79xx phones, its called Cisco::IPPhone and you can find pretty much everything you need within the authors code ... http://search.cpan.org/author/MRPALMER/Cisco-IPPhone-0.05/IPPhone.pm Rgds, Adam -Original Message- From: Marcel Prisi [mailto:[EMAIL PROTECTED] Sent: 11 September 2003 10:00 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco 7940/7960 XML application hint * 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] Question about cdr_sql fields
Sure is, you can set the accountcode=13213 within each entity of sip.conf (or iax.conf I believe). -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: 04 September 2003 17:10 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Question about cdr_sql fields Hello- Is it possible to set the CDR record field called accountcode from within the dialplan? Or is there another way to cause this field to be set, preferably without using AGI code. Thanks Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England www.evtmedia.com ___ 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] DTMF tones not long enough on out going calls
Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? -Original Message- From: James Sizemore [mailto:[EMAIL PROTECTED] Sent: 22 August 2003 17:33 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] DTMF tones not long enough on out going calls DTMF tones are not long enough on out going calls, when I'm using either info or rfc2833. Does anyone know if the tone length value is in rtp.c or chan_sip.c ? ___ 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
[Asterisk-Users] Cisco 79xx XML carriage returns/line feeds
Hi All, I've been developing all sorts of applications for use on our 79xx handsets but am having great difficulty with formatting, I just can't seem to be able to produce a line feed between lines on the stuff actually displayed on the phone. Has anyone else has experience or success with this ? Cheers, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_mysql
I'm not running the latest CVS release but found a couple of days ago that CDR's were not being inserted into my MySQL tables, I restarted Asterisk and it worked fine again ... -Original Message- From: Tais M. Hansen [mailto:[EMAIL PROTECTED] Sent: 18 August 2003 18:09 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cdr_mysql -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Is cdr_mysql broken in latest CVS? It builds and loads fine but it doesn't insert cdrs in the database and there's no debug output at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QPoZ2TEAILET3McRAsEyAKCSZFgFSNvweA9Lh1BW1FJFwTwJNACdFNN3 tFLJlAxupabP17gRrVL0VJA= =3k4Y -END PGP SIGNATURE- ___ 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] Malicious Call Trace
I didn't get any feedback on this, I guess its nobody else has come across the requirement maybe ? -Original Message- From: Low, Adam [mailto:[EMAIL PROTECTED] Sent: 12 August 2003 12:29 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Malicious Call Trace All, Has anyone had any thoughts/discussion on providing a malicious call trace feature within Asterisk. Most legacy PBX's support this feature which allows a handset user to indicate using DTMF during a call that it's a malicious call which instructs the PBX to send a specific Q931 message over the ISDN to the providers switch telling it to log the call details as malicious for later reference or blocking. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * 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
[Asterisk-Users] Malicious Call Trace
All, Has anyone had any thoughts/discussion on providing a malicious call trace feature within Asterisk. Most legacy PBX's support this feature which allows a handset user to indicate using DTMF during a call that it's a malicious call which instructs the PBX to send a specific Q931 message over the ISDN to the providers switch telling it to log the call details as malicious for later reference or blocking. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Brenton, Yves, ... I've located the cause of the problem in chan_sip.c but am still trying to find the exact cause being completely new to the asterisk code. It seems that there was an added function in 1.135 called 'find_user' that is supposed to lookup the users incoming call limit but the routine is unable to find a matching user for my AS5300 which I suspect is because it does not REGISTER with the server prior to attempting to send calls. I'm going to continue debugging a little later and see if I can narrow it down more ... Adam -Original Message- From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: 30/07/03 14:09 Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 Hi, I am using the latest cvs release of asterisk, and the behaviour is in fact the same, outbound calls work fine, but for inbound calls (from C2651 over PSTN) , SIP messages get blocked by asterisk, and never reach the phone. The setup is the same : 7960 -- asterisk -- C2651- PSTN Yves |-+- | | Low, Adam | | | [EMAIL PROTECTED]| | | Sent by: | | | [EMAIL PROTECTED]| | | .digium.com | | | | | | | | | 30/07/2003 11:37 | | | Please respond to | | | asterisk-users| | | | |-+- --- | | | | To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] | | cc: | | Subject: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134 | --- | All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP- Asterisk -SIP- AS5300 But inbound calls fail, I see the initial INVITE from the AS5300 which is received by asterisk but not responded to and then the AS5300 sends another few INVITE's which are received but ignored assumable as they were duplicates for the first. Unfortunately since I've been trying the different cvs revisions of chan_sip.c I've got susbequent problems with the server crashing after the first INVITE from the AS5300 using anything greater than cvs 1.134 I suspect this is something to do with the per-user limits added in cvs 1.135 but I am curious to see if anyone has any problems with the latest cvs elease of asterisk with SIP ? Adam Sip read: INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0 Via: SIP/2.0/UDP 213.160.252.50:53893 From: 611012210 sip:[EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown Date: Wed, 30 Jul 2003 09:26:11 GMT Call-ID: [EMAIL PROTECTED] Cisco-Guid: 1667049428-3407675953-0-149543808 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 1059557171 Contact: sip:[EMAIL PROTECTED]:5060;user=phone Expires: 180 Content-Type: application/sdp Content-Length: 149 v=0 o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50 s=SIP Call c=IN IP4 213.160.252.50 t=0 0 m=audio 20032 RTP/AVP 8 0 65535 18 15 headers, 6 lines Using latest request as basis request Sending to 213.160.252.50 : 53893 (non-NAT) Found audio format 8 Found audio format 0 Found audio format 65535 Found audio format 18 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 AM00CM01*CLI Disconnected from Asterisk server * 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may
RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
Well found Patrick, that did the trick for me as well !
I had been trying to debug 1.135 where this portion of code wasn't added yet ... thats
a lesson learnt ...
-Original Message-
From: Patrick
To: '[EMAIL PROTECTED] '
Sent: 30/07/03 15:04
Subject: RE: [Asterisk-Users] chan_sip.c problems problems from cvs 1.134
It is in the find_user() routine. If it is not an extension on the
PBX,
it should return a zero
if ( isfound ) {
ast_log(LOG_DEBUG, %s is not a local user\n, name);
ast_pthread_mutex_unlock(userl.lock);
return 1; --- this is the problem - change it to a 0.
}
It isn't an error, so it should just return. Change that and the
function
will work properly. I tested it using an AS5350 and successly made an
inbound call.
Patrick
On Wed, 30 Jul 2003, Low, Adam wrote:
Brenton, Yves, ...
I've located the cause of the problem in chan_sip.c but am still
trying to find the exact cause being completely new to the asterisk
code. It seems that there was an added function in 1.135 called
'find_user' that is supposed to lookup the users incoming call limit but
the routine is unable to find a matching user for my AS5300 which I
suspect is because it does not REGISTER with the server prior to
attempting to send calls.
I'm going to continue debugging a little later and see if I can narrow
it down more ...
Adam
-Original Message-
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: 30/07/03 14:09
Subject: Re: [Asterisk-Users] chan_sip.c problems problems from cvs
1.134
Hi,
I am using the latest cvs release of asterisk, and the behaviour is in
fact
the same,
outbound calls work fine,
but for inbound calls (from C2651 over PSTN) , SIP messages get
blocked
by asterisk, and never reach the phone.
The setup is the same : 7960 -- asterisk -- C2651-
PSTN
Yves
|-+-
| | Low, Adam |
| | [EMAIL PROTECTED]|
| | Sent by: |
| | [EMAIL PROTECTED]|
| | .digium.com |
| | |
| | |
| | 30/07/2003 11:37 |
| | Please respond to |
| | asterisk-users|
| | |
|-+-
---
|
|
|
| To: '[EMAIL PROTECTED]'
[EMAIL PROTECTED] |
| cc:
|
| Subject: [Asterisk-Users] chan_sip.c problems problems from
cvs 1.134 |
---
|
All,
I've found problems in my setup with the latest couple of revisions
(1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9
asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's,
everything
is in the same VLAN and only running SIP.
Outbound calls work fine: 7940 -SIP- Asterisk -SIP- AS5300
But inbound calls fail, I see the initial INVITE from the AS5300 which
is
received by asterisk but not responded to and then the AS5300 sends
another
few INVITE's which are received but ignored assumable as they were
duplicates for the first.
Unfortunately since I've been trying the different cvs revisions of
chan_sip.c I've got susbequent problems with the server crashing after
the
first INVITE from the AS5300 using anything greater than cvs 1.134
I suspect this is something to do with the per-user limits added in
cvs
1.135 but I am curious to see if anyone has any problems with the
latest
cvs elease of asterisk with SIP ?
Adam
Sip read:
INVITE sip:[EMAIL PROTECTED];user=phone;phone-context=unknown SIP/2.0
Via: SIP/2.0/UDP 213.160.252.50:53893
From: 611012210 sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED];user=phone;phone-context=unknown
Date: Wed, 30 Jul 2003 09:26:11 GMT
Call-ID: [EMAIL PROTECTED]
Cisco-Guid: 1667049428-3407675953-0-149543808
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled
CSeq: 101 INVITE
Max-Forwards: 6
Timestamp: 1059557171
Contact: sip:[EMAIL PROTECTED]:5060;user=phone
Expires: 180
Content-Type: application/sdp
Content-Length: 149
v=0
o=CiscoSystemsSIP-GW-UserAgent 5241 1607 IN IP4 213.160.252.50
s=SIP Call
c=IN IP4 213.160.252.50
t=0 0
m=audio 20032 RTP/AVP 8 0 65535 18
15 headers, 6 lines
Using latest request as basis request
Sending to 213.160.252.50 : 53893 (non-NAT)
Found audio format 8
Found audio format 0
Found audio format 65535
Found audio format 18
Capabilities: us - 524302, them - 268/0, combined - 12
Non-codec
RE: [Asterisk-Users] RTP session traversing Asterisk server ...
Thanks all, I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections. I don't seem to be hitting the bug that Dave mentioned below ... -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 its a know problem. I have played with the canreinvite stuff to no end and have never gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if it does P2P with my Cisco Phones then it might just be a message thing on * server. Dave Packham [EMAIL PROTECTED] 7/28/2003 4:16:16 PM On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment. - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 28, 2003 11:29 AM Subject: [Asterisk-Users] RTP session traversing Asterisk server ... I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points. Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ? Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 * 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] stupid questions ..
1. what's the sequence to press on a SIP phone to transfer a call to another extension. Which SIP phone? Soft/hard ? Phone specific ... 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? Which SIP phone? Soft/hard ? Phone specific ... 3. what's the extensions.conf syntax to dial two SIP extensions at once? Separate the dial peer with a as follows: exten = 13646,1,Dial(SIP/4840SIP/4841) many thanks Dave ___ 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] stupid questions ..
You got it, I have cisco 7940 phones which have a transfer soft key which tells the phones SIP UA to transfer the call via Asterisk to another SIP UA ... -Original Message- From: Dave Alan Caruana [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 13:26 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] stupid questions .. Sip phones on the system are Grandstream Budgettone 100's. Was assuming it wouldn't be phone specific :) they have flash key which is meant to send a DTMF. thanks for the help with the dial string. Dave - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 29, 2003 11:28 AM Subject: RE: [Asterisk-Users] stupid questions .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. Which SIP phone? Soft/hard ? Phone specific ... 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? Which SIP phone? Soft/hard ? Phone specific ... 3. what's the extensions.conf syntax to dial two SIP extensions at once? Separate the dial peer with a as follows: exten = 13646,1,Dial(SIP/4840SIP/4841) many thanks Dave ___ 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 ___ 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] Linux flavor?
Personally, I've compiled Asterisk on Redhat and Debian without any problems on either, I think generally Asterisk compiles very easily no matter what the distro but I would recommend that you use the one you are most comfortable/experienced with. -Original Message- From: Sean Rodger [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 15:02 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Linux flavor? What Linux distribution is best for use with Asterisk? (easiest compile, least problems, etc) Thanks, Sean Rodger ___ 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] RTP session traversing Asterisk server ...
Sure, nothing special though: [4840] type=friend username=4840 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband [4842] type=friend username=4842 host=dynamic canreinvite=yes nat=no qualify=200 mailbox=4840 dtmfmode=inband -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 15:43 To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server ... can you share the SIP conf entries that you are using to get this to work? I have played with the canreinvite and reinvite entries but cannot make my 7960's do P2P I am running the 5.1 SIP code on the phones. Dave [EMAIL PROTECTED] 7/29/2003 3:13:54 AM Thanks all, I spent some time on this last night with packet sniffer in hand, the 'canreinvite' option makes sense and seems to work well for me (running latest * CVS release) when used between 79xx phones and the AS5300 gateway although I get some somewhat expected problems with 79xx that are NAT'd behind ADSL/cable connections. I don't seem to be hitting the bug that Dave mentioned below ... -Original Message- From: Dave Packham [mailto:[EMAIL PROTECTED] Sent: 29 July 2003 04:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server ... Check out this bug http://bugs.digium.com/bug_view_page.php?bug_id=005 its a know problem. I have played with the canreinvite stuff to no end and have never gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if it does P2P with my Cisco Phones then it might just be a message thing on * server. Dave Packham [EMAIL PROTECTED] 7/28/2003 4:16:16 PM On your sip.conf for each sip endopoint set canreinvite = yes. That way the rtp stream won t go through *. The only problem though is for ATA 186. They need canreinvite = No when they are in a NAT environment. - Original Message - From: Low, Adam [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 28, 2003 11:29 AM Subject: [Asterisk-Users] RTP session traversing Asterisk server ... I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the server and ultimately a higher latency between the two end points. Is this a typical operation of Asterisk or is this possibly due to the fact that some of the phones (not those used in the tests) are running NAT and Asterisk relays all RTP packets ? Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 * 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 * 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
RE: [Asterisk-Users] can't get musiconhold to work
I've not got a sound card in my RH9 * box and music on hold works great as long as you have mpg123 in /usr/bin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 27 July 2003 20:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] can't get musiconhold to work Yes always end your conf files with blank lines otherwise you may get weird results from asterisk.. as for the sond card requirement I don't know all my systems have onboard sound.. So you mean a just simple blank line at the end of the musiconhold.conf file or the extensions.conf file? Second question, though it might seem a bit stupid, do I perhaps need a sound card on the box that asterisk is running on? I don't think this should be the case but I'm just wondering. Is there anything I can do to manually make it run with asterisk? I guess what I'm trying to say here is in ps aux I see no example of mpg123 running that tells me it has not been executed. What is the process that asterisk uses to execute it? Is it executed each time a caller is put on hold or are instances started in the background when asterisk begins (listen state)? AJ On Sat, 26 Jul 2003, WipeOut . wrote: Only things I can suggest is.. 1. Execute it from a command line and make sure it runs.. If not you may hevr to compile it from source.. 2. Make sure you have a new line at the end of your .conf file cos * often freaks out about that.. Other than that I don't know why its not working for you.. No instances of it running when I look at processes. AJ On Sat, 26 Jul 2003, WipeOut . wrote: Sorry I though you had compiled from source... When * is running do ps-aux | grep mpg123 and make sure it is actually running.. Later.. Wipeout I'm using the exact mpg123 binary that you sent me. When I execute a whereis mpg123 it returns /usr/bin. To take it a step further I've done whereis mpg321 and rpm -q mpg321 just to make sure mpg321 is not on the system. The one thing that's confusing the heck out of me is the fact that the rpm that I installed seems to have installed in /usr/bin whereas everybody else's installed in /usr/local/bin. Any other ideas? I'm growing very frustrated. AJ On Sat, 26 Jul 2003, WipeOut . wrote: IIRC I had the same problem becasue the package will install the mpg123 binary to /usr/local/bin and * seems to look in /usr/bin so just copy the mpg123 executable to /usr/bin and it should work.. Later.. I can't seem to get musiconhold to work. I'm running asterisk on a RH9 box, I have the mpg123 package installed. In my zapata.conf file I have the line MusicOnHold=default . In my musiconhold.conf file, in the classes section I uncommented default and loud. In my extensions.conf file I have a set musiconhold line. However if I get a call and I either put it on hold or hit flash I get no music. The sample mp3 file is in the mohmp3 directory. Does anyone know what I might be doing wrong or how I might be able to correct it? Also I have tried assigning a extension with the MusicOnHold application and it still doesn't seem to work. AJ ___ 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 -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ 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
RE: [Asterisk-Users] Dialogic hardware
I asked the same question a couple of weeks ago and was told by Digium that its not commercially available yet but the source code is available under NDA with Digium. I'll dig out my contact and send off-list ... Adam -Original Message- From: Marcel Prisi [mailto:[EMAIL PROTECTED] Sent: 25 July 2003 11:30 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dialogic hardware Hi all ! What is the current status of the Dialogic channel driver ? Is it available ? Is it commercial ?? Any info ? Thanks ___ 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
[Asterisk-Users] 7940 AS5300 codec issues/questions G.729 G.711
I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure which interoperate the best with each other and so was wondering if anyone call tell me if they have similar setups with this working and if so which codec they choose for the AS5300 ... When comparing the G711alaw against G729a I did not expect to have so many breaks in the sound when using G729a and wondered if others had experienced this. I would expect G729 to be a lesser overall quality and sampling rate but not to effectively lose some speech altogether ... * 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] SIP Call Forwarding/Transfer support ?
Hielke John, I too have the 7940 phones working perfectly in my setup with the exception of 7940's that are NAT'd when SIP'ing towards the AS5300 and then I find one way voice path but strongly suspect the mini firewall's we are using but am yet to debug this. John has some example configs on his site which were a good help in getting me started and am sure will help you: http://www.loligo.com/asterisk/ John, you mentioned transfers and this is my obstacle at the moment, could you share any more insight to your setup on this ? Adam -Original Message- From: John Todd To: [EMAIL PROTECTED] Sent: 23/07/03 10:08 Subject: Re: [Asterisk-Users] SIP Call Forwarding/Transfer support ? Hielke - Cisco 7960 phones work quite well with Asterisk and SIP, and I have been using them for many months now, both on my own systems and those of my clients. Perhaps you can forward your configurations and I can help debug. The only functions I have not had working 100% are: parking via the transfer button on the phone, and peer-to-peer SIP calls (I have to set canreinvite=no on the config settings for the phones) JT Hello Adam, i am doing some testing in the same direction. I want to use Asterisk with Cisco 7960, Grandstream and Pingtel SIP phones. For receiving and terminating calls i want to use the Nikotel SIP service. Until now i had no luck in getting the Cisco 7960 phone to work. But with the Budgetone Grandstream phone i could receive and make calls. Blind call forwarding also worked. I think the reason for the 7960 not working is a bug in the Asterisk. Something with a wrong Cseq. Regards, Christian. -- Hielke C. Braun VP system engineering * 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
[Asterisk-Users] Asterisk - SIP - AS5300 signalling missing on connect/clear call
Hi All, I seem to be having a problem with calls from Asterisk into the AS5300, I am sniffing the session between the AS5300 and the Asterisk server and I see the Asterisk server send a SIP INVITE and the AS5300 responds with a SIP 100 TRYING but then I do not see any more SIP signalling messages from the AS5300 once the call connects or clears on the ISDN side. Has anyone else experienced similar problems ? Finally I do a clear on the 7960 SIP phone and the call gets cleared. Calling in the opposite directions works perfectly ... Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - AS5300 SIP Interoperability
Greetings, I am attempting to configure an AS5300 to provide a SIP based gateway to the PSTN from Asterisk. I have been unable to identify through the docs how specifically this should be configured in Asterisk and have not been able to get things working through trial and error. I am sure I am missing something fairly obvious here but any guidance (or example cfgs) would be much appreciated. Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability
Thanks Daniel Gustavo, I had the AS5300 configured ok and could make calls PSTN AS5300 ASTERISK 7940 no problem but outbound from Asterisk to the AS5300 wasn't working ... until now (wasn't sure about the sip.conf) ... thanks again gents ! -Original Message- From: Daniel Concepcion [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:40 To: [EMAIL PROTECTED]; Low, Adam Subject: Re: [Asterisk-Users] Asterisk - AS5300 SIP Interoperability * 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] Cisco 7960
Title: Message William, I am running 7960/7940's with 5.1 (Asterisk SIP) without problems although I did have some issues (too numerous to mention)with new phones that had never been operated on a CallManager network first. It seems the firmware must be upgraded to support SIP and this can only be done with CallManager (apparently). The only way I managed to figure everything out was with a packet analyser, I don't suppose you have the possibility of doing that ? Rgds, Adam -Original Message-From: William Carlson [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 13:40To: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Cisco 7960 lol well I probaly should ask a question lol. Any idea what could be causing this? Also I cannot call from my pingtel phone to the 7960 but I can call the other way around. any ideas on that? Thanks, Will - Original Message - From: William Carlson To: [EMAIL PROTECTED] Sent: Thursday, July 17, 2003 7:34 AM Subject: [Asterisk-Users] Cisco 7960 I bought a 7960 it was running version 3.3 of the SIP software. It worked fine. Me being the idiot I am upgraded to 5.1. Now it downloads the configs and then reboots. if I unplug the ethernet it doesn't rebootor if I remove all the lines in the SIP config it won't reboot. Since this is used cisco won't give me any support. For now I am running the MGCP version but eh asterisk seems to have some issues with it. Thanks, Will * 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
RE: [Asterisk-Users] Help Needed
Not me I'm afraid, I'm running Asterisk -SIP- Cisco AS5300 -E1- PSTN .. no Quicknet hardware for me ... -Original Message- From: Arun Kumar Sharma, Noida [mailto:[EMAIL PROTECTED] Sent: 17 July 2003 15:49 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Help Needed Thanks Adam, This document provides me a high level architecture of Asterisk. Can you please tell me if I want to evaluate Asterisk on an Intel PC which Quicknet hardware will be required to just run a POTS to SIP call? Thank you once again for very fast response. Regards Arun * 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] Firewall Silly - anyone can help with a CVS tar ball ?
James, thanks I appreciate it. -Original Message- From: James Golovich To: '[EMAIL PROTECTED]' Sent: 17/06/03 18:41 Subject: Re: [Asterisk-Users] Firewall Silly - anyone can help with a CVS tar ball ? * 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
