Re: [asterisk-users] Quiet 24 port POE gig switch
Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange Packet Behavior
Hello Everybody! My server is attempting to connect to a SIP device, but is not succeeding to. I checked the actual packets traveling back and forth with ngrep and I noticed some odd packets coming in. Here is the outgoing INVITE packet sent to the SIP device: # U asteriskserver:5060 - sipdevice:15244 INVITE sip:3...@sipdevice:15244 SIP/2.0..Via: SIP/2.0/UDP asteriskserver:5060;branch=z9hG4bK7cb4b7a7;rport..From: m sip:6571...@asteriskserver;tag=as6e2a8098..To: sip:3...@sipdevice:15244..Contact: sip:6571...@asteriskserver..Call-ID: 5f79e4897d6ab4793ed0724935192...@asteriskserver..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Sun, 01 Feb 2009 10:27:14 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: application/sdp..Content-Length: 269v=0..o=root 17661 17661 IN IP4 asteriskserver..s=session..c=IN IP4 asteriskserver..t=0 0..m=audio 11272 R TP/AVP 18 101..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv.. Unfortunately, there is no response from the sip device. However, this packet shows up with ngrep from the sipdevice: # I sipdevice - asteriskserver 3:1 E...5...INVITE sip:6571...@sipdevice SIP/2.0..Via: SIP/2.0/UDP asteriskserver:5060;branch=z9hG4bK 0102c4c0;rport..From: m sip:6571...@asteriskserver;tag=as47be5d18..To: sip:3...@216.235.152.22..Contact: sip:6571...@asteriskserver..Call-ID: 16e6211c310d36905520dcb625f86...@asteriskserver..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max -Forwards: 70..Date: Sun, 01 Feb 2009 10:36:51 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Supported: replaces..Content-Type: a Apparently, the device is getting a packet, but its response is not being sent back properly. The sipdevice is sending a packet, but instead of being sent to port 5060, it is sent an odd port of 3:1. Moreover, there are some phantom characters that show up the beginning of the return packet (E...5...I). Any help would be great! Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some information on SS7 parameters
Hello List I am setting up a small demo site using SS7 and one of the requirement is to be able to unhide the numbers and locate exact location of the caller (BTS ID). Vodafone uses Nokia-Siemens switch and has confirmed that the parameters will be sent to the us. I just want to know how do read those information from the dialplan to be able to present them to the Agent Thanks Samwel Muro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax clients were unregistered after 30sec
by using rtcachefriends=yes it was done. --- On Sat, 1/31/09, Pezhman Lali pezhman_l...@yahoo.com wrote: From: Pezhman Lali pezhman_l...@yahoo.com Subject: [asterisk-users] iax clients were unregistered after 30sec To: asterisk-users@lists.digium.com Date: Saturday, January 31, 2009, 7:34 PM Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language | mailbox | deny | permit | qualify | disallow | allow | ipaddr | port | regseconds | email |date | user_id -+--++--+---+--+--++-+--+-+--+--+-+---+-+--+-+--++-+--+---+-+--+++-+- 9706015 | 9706015 | friend | 5056ed3c | | | no || | md5 | | | 9706015 | GPHONE | | dynamic | | | || yes | NULL | all | 0.0.0.0 |0 | 0 | pezhman_l...@yahoo.com | 2009-01-31 11:33:10 | 9706015 (1 row) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
Beware PoE switches that can't handle Class 3 (15W) on all ports. Most have fans because 24 (or 48) x 15W is hot! IanC -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: Sunday, February 01, 2009 11:58 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ Moffat Communications Limited is a wholly owned Company of Grant Thornton UK LLP, registered in England and Wales, with number 2355810, and whose registered office is Grant Thornton House, Melton Street, Euston Square, London, NW1 2EP. Moffat Communications Limited is registered to carry out Business and management consultancy. When addressed to our clients, any opinions or advice set out in this email and any attachments are subject to the terms and conditions expressed in the relevant Moffat Communications Limited client engagement letter. Opinions, conclusions and other information in this email which do not relate to the official business of the firm are neither given nor endorsed by it. Any personal views expressed in this message are not necessarily the views of Moffat Communications Limited. This message is intended only for the individual or entity to which it is addressed and may contain information that is private and confidential. If you are not the intended recipient, employee or agent responsible for delivering the message to the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited. If you have received this communication and its attachments in error, please return the original message and attachments to us using the reply facility on e-mail. Moffat Communications Limited has scanned this email for viruses but does not accept any responsibility once this email has been transmitted. You should scan attachments (if any) for viruses. Any person communicating with us by e-mail will be deemed to have accepted the risks associated with sending information by e-mail being interception, amendment and loss, and also the consequences of incomplete or late delivery. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
2009/1/31 OCG Technical Support supp...@ocg.ca A little off topic but I need to put a 24 port Gig PoE switch into a small office – no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I can't remember the exact model but I used a Cisco catalyst switch for a class and after reboot, it was very quiet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 109
Sorry, but why u r using the Radius with the CDR? Not enough to access the CDR in the /var/log/asterisk/cdr-csv/Master.csv? Also, what kind of Radius u r using? Any suggested link? Regards Bilal Hello list. I'm having some problems with the CDR Radius in my Asterisk 1.4. I'm using two TC400B cards for transcoding. When I reach nearly 100 simmultaneous calls, the CDR radius packets are being duplicated and I'm getting this message in the asterisk console : cdr_radius.c:227 radius_log: Failed to record Radius CDR record! I'm also using the radiusclient-ng 0.5.6 to interact with the radius server. What I'm seeing in the radius server side are 3 radius packets reaching at almost the same time. Please look at the debug: Fri Jan 30 23:34:08 1998: DEBUG: Packet dump: *** Received from 10.10.149.211 port 56844 Code: Accounting-Request Identifier: 129 Authentic: 237U254W129156159D153A$209233`G9 Attributes: Acct-Status-Type = Stop Asterisk-Acc-Code = TO_PROVIDER Asterisk-Src = 550272 Asterisk-Dst = 0130005411234618 Asterisk-Dst-Ctx = INC_CALLS Asterisk-Clid = sipp 550272 Asterisk-Chan = SIP/5060-0972ae58 Asterisk-Dst-Chan = SIP/TO_ITSP1-0972f830 Asterisk-Last-App = Dial Asterisk-Last-Data = SIP/005411234...@to_itsp1 Asterisk-Start-Time = 2009-01-30 14:51:23 + Asterisk-Answer-Time = 2009-01-30 14:51:28 + Asterisk-End-Time = 2009-01-30 14:52:50 + Asterisk-Duration = 87 Asterisk-Bill-Sec = 82 Asterisk-Disposition = ANSWERED Asterisk-AMA-Flags = DOCUMENTATION Asterisk-Unique-ID = 1233327083.102 Asterisk-User-Field = B2BUA-PROVIDER-PRUEBAS User-Name = SIP/5060-0972ae58 Acct-Session-Id = 1233327083.102 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 10.10.149.211 Fri Jan 30 23:34:08 1998: DEBUG: Handling request with Handler 'Request-Type = Accounting-Request' Fri Jan 30 23:34:08 1998: DEBUG: Deleting session for SIP/5060-0972ae58, 10.10.149.211, 0 Fri Jan 30 23:34:08 1998: DEBUG: Handling with AuthINTERNAL: Fri Jan 30 23:34:08 1998: DEBUG: AuthBy INTERNAL result: ACCEPT, Fixed by DefaultResult Fri Jan 30 23:34:08 1998: DEBUG: Accounting accepted Fri Jan 30 23:34:08 1998: DEBUG: Packet dump: *** Sending to 10.10.149.211 port 56844 Code: Accounting-Response Identifier: 129 Authentic: 237U254W129156159D153A$209233`G9 Attributes: Fri Jan 30 23:34:08 1998: DEBUG: Packet dump: *** Received from 10.10.149.211 port 56844 Code: Accounting-Request Identifier: 129 Authentic: 237U254W129156159D153A$209233`G9 Attributes: Acct-Status-Type = Stop Asterisk-Acc-Code = TO_PROVIDER Asterisk-Src = 550272 Asterisk-Dst = 0130005411234618 Asterisk-Dst-Ctx = INC_CALLS Asterisk-Clid = sipp 550272 Asterisk-Chan = SIP/5060-0972ae58 Asterisk-Dst-Chan = SIP/TO_ITSP1-0972f830 Asterisk-Last-App = Dial Asterisk-Last-Data = SIP/005411234...@to_itsp1 Asterisk-Start-Time = 2009-01-30 14:51:23 + Asterisk-Answer-Time = 2009-01-30 14:51:28 + Asterisk-End-Time = 2009-01-30 14:52:50 + Asterisk-Duration = 87 Asterisk-Bill-Sec = 82 Asterisk-Disposition = ANSWERED Asterisk-AMA-Flags = DOCUMENTATION Asterisk-Unique-ID = 1233327083.102 Asterisk-User-Field = B2BUA-PROVIDER-PRUEBAS User-Name = SIP/5060-0972ae58 Acct-Session-Id = 1233327083.102 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 10.10.149.211 Fri Jan 30 23:34:08 1998: INFO: Duplicate request id 129 received from 10.10.149.211(56844): ignored Fri Jan 30 23:34:08 1998: DEBUG: Packet dump: *** Received from 10.10.149.211 port 56844 Code: Accounting-Request Identifier: 129 Authentic: 237U254W129156159D153A$209233`G9 Attributes: Acct-Status-Type = Stop Asterisk-Acc-Code = TO_PROVIDER Asterisk-Src = 550272 Asterisk-Dst = 0130005411234618 Asterisk-Dst-Ctx = INC_CALLS Asterisk-Clid = sipp 550272 Asterisk-Chan = SIP/5060-0972ae58 Asterisk-Dst-Chan = SIP/TO_ITSP1-0972f830 Asterisk-Last-App = Dial Asterisk-Last-Data = SIP/005411234...@to_itsp1 Asterisk-Start-Time = 2009-01-30 14:51:23 + Asterisk-Answer-Time = 2009-01-30 14:51:28 + Asterisk-End-Time = 2009-01-30 14:52:50 + Asterisk-Duration = 87 Asterisk-Bill-Sec = 82
[asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
I have the same issue, I was just asking about that. My main SIP PRI provider identifies me from my IP address, but I have two separate PRIs (different rate centers) with them. From their end I get calls to xxx.yyy.zzz.xxx and .xxy and I have no trouble getting calls, but I can only send calls from my main machine IP address so I can't control where I am sending calls to. I am hoping to have this developped somehow (a per SIP peer bindaddr and bindport), even if it means some bounty. I can't imagine this being this difficult, so a few of us who need this putting a couple hundred dollar would probably do it. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 12:45 To: bilmar...@yahoo.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
As far as I know all POE switches are quite noisy, they need to cool the extra power consumed by the POE and hence they will run warmer than other switch. I know Cisco, 3COM, are very noisy but you can try other cheaper brand like levelone or other to see if they have fans inside the switch. Good Luck -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Monday, February 02, 2009 1:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Quiet 24 port POE gig switch 2009/1/31 OCG Technical Support supp...@ocg.ca A little off topic but I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I can't remember the exact model but I used a Cisco catalyst switch for a class and after reboot, it was very quiet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
I guess one would have to ask whether 1000 Gb is necessary. That's a lot of bandwidth. It might make sense to use it for central distribution. There are also some that have one or two 1000 Gb ports that might be appropriate for trunking and the rest 100 Mb which is probably fast enough for terminal nodes. That combination would be less power hungry. On the other hand if this is for HDTV multichannel distribution, then I retract what I said. Wilton ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
On Sunday 01 February 2009 11:32:51 bilal ghayyad wrote: I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? You cannot. This behavior is not supported, even in trunk. What is currently supported in 1.6.0 and above, however, is responding back on the same IP, if the opposing SIP server started the request. This was necessary to support TCP. It may no longer be that difficult to support specifying a source IP per peer, given the changes necessary to support TCP, but as I have not attempted it, I'll hold off on pronouncing that an easy change. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Apologies, I tend to give the impression that I am sure changes are easy: I was merely giving an educated guess as a programmer but not specifically as an Asterisk or even Linux programmer. I definitely could be way off in my evaluation of the work involved. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Sunday, February 01, 2009 14:25 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine On Sunday 01 February 2009 11:32:51 bilal ghayyad wrote: I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? You cannot. This behavior is not supported, even in trunk. What is currently supported in 1.6.0 and above, however, is responding back on the same IP, if the opposing SIP server started the request. This was necessary to support TCP. It may no longer be that difficult to support specifying a source IP per peer, given the changes necessary to support TCP, but as I have not attempted it, I'll hold off on pronouncing that an easy change. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
At the risk of seeming impolite (I really am not), why not? Isn't Asterisk able to send packets using another interface using bindaddr? The problem, for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 14:56 To: bilal ghayyad Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Could you not use some iptables to do this? I don't know the exact command you'd need but it could work something like... If the destination port is 5060 and destination ip is xxx then route via the default ip (so do nothing) If the destination port is 5061 and destination ip is xxx change the destination port back to 5060 and set secondary ip as the source? Just a thought... i'm guessing this would be able to do the job.. not sure what issues you might run in to by changing 5060 to 5061... but if it came to it you could try it by using an alternate ip and changing it back. Who knows... not sure if i've even read enough to understand the problem :) Cheers Geraint 2009/2/1 Mike l...@virtutel.ca At the risk of seeming impolite (I really am not), why not? Isn't Asterisk able to send packets using another interface using bindaddr? The problem, for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 14:56 To: bilal ghayyad Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
[asterisk-users] iChat voice (and maybe video?)
Hi Dudes, i searched for some time for an answer for this, i found some posting from John Todd half a decade ago [1], was there some chance in this? Is it somehow possible to voip from ichat to asterisk? If there's no light, is this something that could happen with enough founding, or is Mapple preventing this somehow (legal or technical)...? Regards, Andreas [1] http://lists.digium.com/pipermail/asterisk-dev/2003-July/001075.html _ Time for a change? SEEK and you shall find. http://a.ninemsn.com.au/b.aspx?URL=http%3A%2F%2Fmsn%2Eseek%2Eco%2Enz%2FID%5FSEEKNZMAIN%5FUSR%2FPages%2Falliance%5Fhomepage%2Eascx%3FComeFrom%3Dmsnnz%26tracking%3Dsk%3Atl%3Asknz%3Amsnnz%3A0%3Ahottag%3Aflirt_t=757263783_r=SEEKNZ_tagline_m=EXT___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
I briefly glanced at the code before responding, and it does seem that if you specify a bind address it will use that address when responding. I stick by my comment that the change you want is not exactly simple - unless you are very familiar with the 18,000 line chan_sip.c :) It also appears that the bind address is a global in this implementation... so some thought would have to go into efficiently representing all the possible peer bind addresses in a thread safe manner... and if the change is made to this channel it really out to be for all of them. So a hack is probably possible without too much trouble. For it to be done right is not a simple change. Sorry for the rambling. j On Sun, 1 Feb 2009, Mike wrote: At the risk of seeming impolite (I really am not), why not? Isn't Asterisk able to send packets using another interface using bindaddr? The problem, for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 14:56 To: bilal ghayyad Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Actually I think that is a good idea. In sip.conf setup the two remote ends on different IPs (one of which is actually bogus). Outbound NAT based on the destination, where you change the source IP to the one expected by the provider, and change the bogus destination to the real one. Inbound NAT back to the base address based on the destination in the reply. Now THAT is a hack. j On Sun, 1 Feb 2009, Geraint Lee wrote: Could you not use some iptables to do this? I don't know the exact command you'd need but it could work something like... If the destination port is 5060 and destination ip is xxx then route via the default ip (so do nothing) If the destination port is 5061 and destination ip is xxx change the destination port back to 5060 and set secondary ip as the source? Just a thought... i'm guessing this would be able to do the job.. not sure what issues you might run in to by changing 5060 to 5061... but if it came to it you could try it by using an alternate ip and changing it back. Who knows... not sure if i've even read enough to understand the problem :) Cheers Geraint 2009/2/1 Mike l...@virtutel.ca At the risk of seeming impolite (I really am not), why not? Isn't Asterisk able to send packets using another interface using bindaddr? The problem, for the two of us, is that bindaddr is Asterisk-wide, and not per-peer. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Sunday, February 01, 2009 14:56 To: bilal ghayyad Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine Ah, that makes more sense. Asterisk binding to another IP is not the issue, actually, and even running another instance will not do what you need. Your problem is that the OS itself will stamp outbound packets with the main source IP of the main interface. Asterisk could be modified to send packets with specific IP source, but I don't think that would be a simple change. j On Sun, 1 Feb 2009, bilal ghayyad wrote: OK, if I send for my provider (the destination), it will authenticate based on the IP ONLY, this is the provider system. And once authenticated me based on that IP, it will give me all the schema related to this account. Sometimes I need to use another schema for some calls, I am not able until send for the provider from another IP. Did u get what I need? Regards Bilal --- On Sun, 2/1/09, Jeff LaCoursiere j...@jeff.net wrote: From: Jeff LaCoursiere j...@jeff.net Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Sunday, February 1, 2009, 12:44 PM I am confused as to what you are trying to accomplish. Can you be more specific? It seems that you are making this too complicated. You say that the remote end is providing you two SIP trunks that will come from the same IP address. To distinguish them simply have them authenticate with two different usernames. This does beg the question, though, if the endpoint is the same, why have a separate trunk? How about routing the calls based on differing CID? If you can explain the situation more distinctly perhaps an alternate method will present itself. Hard to imagine a real need for binding to multiple local IP addresses on the asterisk side. If you are REALLY stuck on doing it that way, however, how about simply running a second instance of asterisk? You would have to recompile the source to read config from a second tree, but then your second instance could bind to your aliased address. I suppose you could even trunk the two together if the two instances must pass traffic between each other. How odd :) j On Sun, 1 Feb 2009, bilal ghayyad wrote: Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the first IP address and other calls to be authorized with another IP address, ofcourse I have some reason for this. The idea is: how to control the source IP address that I am sending from it to the other side? Can I determine the source IP address of the SIP trunk while I am configuing my SIP section for that connection? What about the bindaddress? Any help? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
On Sunday 01 February 2009 14:39:11 Jeff LaCoursiere wrote: Actually I think that is a good idea. In sip.conf setup the two remote ends on different IPs (one of which is actually bogus). Outbound NAT based on the destination, where you change the source IP to the one expected by the provider, and change the bogus destination to the real one. Inbound NAT back to the base address based on the destination in the reply. Now THAT is a hack. And it probably won't work. SIP is a known protocol which violates layer separation, encoding IP addresses directly into the application layer. Unless your firewall were able to DPI and modify the addresses within the application layer (which may or may not work, depending on whether Asterisk encodes the message with IP addresses or hostnames), then the whole exercise is doomed to fail. One way which does occur to me that will work, if the OP only needed exactly 2 different addresses, would be to set the bindaddr and tcpbindaddr to different addresses, and send TCP signalling for one peer and UDP signalling for the other. Again, this would only work for exactly 2 peers, not for more. The current code uses a separate socket for each of TCP, TLS, and UDP connections, so this would be the maximum possible without any code changes. One could probably use multiple TCP descriptors without a lot of work. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
On Sunday 01 February 2009 15:40:29 Tilghman Lesher wrote: On Sunday 01 February 2009 14:39:11 Jeff LaCoursiere wrote: Actually I think that is a good idea. In sip.conf setup the two remote ends on different IPs (one of which is actually bogus). Outbound NAT based on the destination, where you change the source IP to the one expected by the provider, and change the bogus destination to the real one. Inbound NAT back to the base address based on the destination in the reply. Now THAT is a hack. And it probably won't work. SIP is a known protocol which violates layer separation, encoding IP addresses directly into the application layer. Unless your firewall were able to DPI and modify the addresses within the application layer (which may or may not work, depending on whether Asterisk encodes the message with IP addresses or hostnames), then the whole exercise is doomed to fail. One way which does occur to me that will work, if the OP only needed exactly 2 different addresses, would be to set the bindaddr and tcpbindaddr to different addresses, and send TCP signalling for one peer and UDP signalling for the other. Again, this would only work for exactly 2 peers, not for more. The current code uses a separate socket for each of TCP, TLS, and UDP connections, so this would be the maximum possible without any code changes. One could probably use multiple TCP descriptors without a lot of work. Something like this might work, though: http://asterisk.drunkcoder.com/patches/20090201__multi_ip_chan_sip_bind.diff.txt Disclaimer: untested code. Written for trunk. Will definitely not work on 1.4 and may or may not cleanly apply to 1.6.0. However, based upon my understanding of the code, it's probably very close to what would be needed to support this. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
Ian Cowley i...@moffat.co.uk wrote: Beware PoE switches that can't handle Class 3 (15W) on all ports. Most have fans because 24 (or 48) x 15W is hot! That's the power supplied .. which'd be at the far end of the wire. The efficiency of the PSU plays a big part in the heat dissipation. The push to compact dimensions doesn't help ... a 400W or thereabouts PSU with 24 independent outputs in 1U height? I suppose if the switch were quite deep it could be workable and quiet. The problem isn't simply of being fanless. But being quiet. Preferably below 32 dBA at 1 metres for most offices. You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. -- /\ Bernd Felsche - Innovative Reckoning, Perth, Western Australia \ / ASCII ribbon campaign | Religion is regarded by the common people X against HTML mail | as true, by the wise as false, and by the / \ and postings | rulers as useful. -- Seneca the Younger ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
My memory of a HP procurve (a 2626 PWR from memory) was that it was quite noisy - have they changed? PaulH OCG Technical Support wrote: Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
Bernd Felsche wrote: Ian Cowley i...@moffat.co.uk wrote: Beware PoE switches that can't handle Class 3 (15W) on all ports. Most have fans because 24 (or 48) x 15W is hot! That's the power supplied .. which'd be at the far end of the wire. The efficiency of the PSU plays a big part in the heat dissipation. The push to compact dimensions doesn't help ... a 400W or thereabouts PSU with 24 independent outputs in 1U height? I suppose if the switch were quite deep it could be workable and quiet. The problem isn't simply of being fanless. But being quiet. Preferably below 32 dBA at 1 metres for most offices. You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. So, size does matter after all. :-) 24 x 15W = 360W. Its not that big a supply really, and spread across a 1U case its not that dense a supply. A 360W desktop PC supply can be pretty quiet, so its sad none of the 1U chassis supplies are. Probably if they used a large impeller fan they could get the noise down. I guess they assume these things will be in cupboards or data centres where nobody cares. This is a poor assumption. Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
My google search says fanless... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales Sent: February 1, 2009 6:49 PM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch Importance: High My memory of a HP procurve (a 2626 PWR from memory) was that it was quite noisy - have they changed? PaulH OCG Technical Support wrote: Check out the HP ProCurve Switch 2610-24-PWR -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall Sent: February 1, 2009 6:58 AM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch I can find FANLESS 24 port PoE 10/100 That's an achievement in itself. Can you post details - I have quite a few locations where that might be useful... TIA. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
What are the chances that this can get eventually wrapped in the Asterisk source? If this works, I will definitely consider upgrading to 1.6 before I originally planned to. Regards, Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Sunday, February 01, 2009 18:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine On Sunday 01 February 2009 15:40:29 Tilghman Lesher wrote: On Sunday 01 February 2009 14:39:11 Jeff LaCoursiere wrote: Actually I think that is a good idea. In sip.conf setup the two remote ends on different IPs (one of which is actually bogus). Outbound NAT based on the destination, where you change the source IP to the one expected by the provider, and change the bogus destination to the real one. Inbound NAT back to the base address based on the destination in the reply. Now THAT is a hack. And it probably won't work. SIP is a known protocol which violates layer separation, encoding IP addresses directly into the application layer. Unless your firewall were able to DPI and modify the addresses within the application layer (which may or may not work, depending on whether Asterisk encodes the message with IP addresses or hostnames), then the whole exercise is doomed to fail. One way which does occur to me that will work, if the OP only needed exactly 2 different addresses, would be to set the bindaddr and tcpbindaddr to different addresses, and send TCP signalling for one peer and UDP signalling for the other. Again, this would only work for exactly 2 peers, not for more. The current code uses a separate socket for each of TCP, TLS, and UDP connections, so this would be the maximum possible without any code changes. One could probably use multiple TCP descriptors without a lot of work. Something like this might work, though: http://asterisk.drunkcoder.com/patches/20090201__multi_ip_chan_sip_bind.dif f.txt Disclaimer: untested code. Written for trunk. Will definitely not work on 1.4 and may or may not cleanly apply to 1.6.0. However, based upon my understanding of the code, it's probably very close to what would be needed to support this. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending Calls via SIP trunk from two different IP addresses from same Asterisk Mac hine
On Sunday 01 February 2009 18:23:04 Mike wrote: What are the chances that this can get eventually wrapped in the Asterisk source? Well, this will have to work, first. Second, it would have to be adapted to work for tcp and tls, too. We'd probably put it up on reviewboard and make sure that it passes muster with the other developers. At the earliest, this could make an appearance in 1.6.2. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
On Fri, Jan 30, 2009 at 3:23 PM, Danny Nicholas da...@debsinc.com wrote: The dialplan AFAIK doesn't cover HOLD handling. If you can spare the overhead, you can make a daemon to watch hints and run a script whenever the hint for a line goes to hold and changes from hold to inuse. Just run asterisk –rx core show hints and asterisk –rx core show channels and integrate the 2 outputs. For your purpose, you can probably just use the first command. You should instaed use the AMI and create an event based solution instead of relying on polling via asterisk -rx !... Check out: http://www.voip-info.org/wiki-Asterisk+manager+API http://www.voip-info.org/wiki/view/asterisk+manager+events Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] early dial: asterisk and ATA
On Thu, Jan 29, 2009 at 6:15 PM, Vieri rentor...@yahoo.com wrote: I'm trying to do the same in the SPA8000 units but without any luck. If anyone is doing something similar with this device then I'd appreciate it if you could share your relevant config options (dial pattern, etc.). Not sure about the SPA8000, but the SPA devices I know (phones + 2102 ATA) all have a per line Dial Plan paramenter that will allow you to acheive that behaviour. Check link sys ATA / Phone Admin Guides. -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Managing codecs
Assuming you are using SIP phones and IIRC, you can hint at the codec to be used by setting the SIP_CODEC variable in the dialplan; before Dial()'ing, of course ! :-) I think this is still an area where asterisk needs improvement... Dynamic codec (re) negotiation. Anyone care to correct me ? Cheers, -- exvito ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk with Polocom Video Conferencing Unit
I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya - Asterisk (1.2.30) - External ISDN Network BUT They also have a Polycom VSX 7000 that with some sort of BRI converters that plugs into the Avaya. The Trunk is working well except for Video Conference Calls. The Polocom can receive but not make calls, and the calls that it receives drop out every 5 minutes. Short of telling them to fork out for a BRI service does anyone have any ideas how to rectify the drop-outs? Would 1.4 help? Is there a way to plug the Polycom into the Asterisk server directly? Any help would be much appreciated. -- Cheers, Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
2009/2/2 Steve Underwood ste...@coppice.org Bernd Felsche wrote: Ian Cowley i...@moffat.co.uk wrote: Beware PoE switches that can't handle Class 3 (15W) on all ports. Most have fans because 24 (or 48) x 15W is hot! That's the power supplied .. which'd be at the far end of the wire. The efficiency of the PSU plays a big part in the heat dissipation. The push to compact dimensions doesn't help ... a 400W or thereabouts PSU with 24 independent outputs in 1U height? I suppose if the switch were quite deep it could be workable and quiet. The problem isn't simply of being fanless. But being quiet. Preferably below 32 dBA at 1 metres for most offices. You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need recesses to allow sufficient airflow when the positions above and below are filled. So, size does matter after all. :-) 24 x 15W = 360W. Its not that big a supply really, and spread across a 1U case its not that dense a supply. A 360W desktop PC supply can be pretty quiet, so its sad none of the 1U chassis supplies are. Probably if they used a large impeller fan they could get the noise down. I guess they assume these things will be in cupboards or data centres where nobody cares. This is a poor assumption. The problem is squeezing fans in that can push enough air to keep it cool... For a 1U device, you have only 4.445cm to work with, with a 4mm fan, that would be 2.2mm of space for casing etc above and below, reasonably tight already... A quiet 80mm fan as you may find in a PC PSU that puts out somewhere between 15-20dBA of noise will typically move between 20 and 30 cfm of air... A quiet 120mm fan at the same noise levels would typically move between 30 and 50 cfm of air and a quiet 40mm at those levels would move about 5 cfm of air... Obviously, they aren't using quiet 40mm fans... To get the airflow of the quiet 80mm fans, a 40mm fan has to go very fast and you're looking at noise levels of approx 40-60dBA, not exactly quiet, but, that's not all, even if the fan was silent, forcing the air through the small cramped chassis of a 1U device is going to be noisy... The assumption made when they make these devices is that the vast majority of people will put this kit somewhere out of the way in a likely temperature controlled, reasonably sound insulated environment, with the rest of their hardware that lives hidden from people... These people will likely prefer that kit uses the space as efficiently as possible, so, squeezing as much functionality into as few rack units as possible is important... They have typically made a good assumption in this I would say... Admittedly, people who are planning an office for their first time may more commonly neglect IT hardware, it's requirements (and those of those people around it), from an IT standpoint, it's a significant pain to deal with, but, in most cases I've seen, it's something that's considered very carefully if planning an office in the future... I suspect the lack of larger quieter units in the market is reflective of the much lower demand for these, somewhat specialised devices... On the otherhand, soundproofed rack cabinets that have integrated cooling and look nice/plain enough that they don't scare people in an office should be generic enough that there would, I suspect, be sufficient demand from those that didn't consider IT requirements when fitting out an office to justify making them... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trunk with Polocom Video Conferencing Unit
2009/2/2 Daniel Harper dan...@harper.net.nz I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya - Asterisk (1.2.30) - External ISDN Network BUT They also have a Polycom VSX 7000 that with some sort of BRI converters that plugs into the Avaya. The Trunk is working well except for Video Conference Calls. The Polocom can receive but not make calls, and the calls that it receives drop out every 5 minutes. Short of telling them to fork out for a BRI service does anyone have any ideas how to rectify the drop-outs? Would 1.4 help? Is there a way to plug the Polycom into the Asterisk server directly? Any help would be much appreciated. The Polycom has a number of interface options, quad BRI being one of them... Your Avaya PBX has some BRI modules with the Polycom probably using 4 of the ports on one of those modules able to use 8 channels at the same time... You've not mentioned how the Avaya is connected to Asterisk and you've not provided any information on messages you are getting on the Asterisk console when attempting to make outbound calls or when calls drop out on inbound calls... This would be useful to help you determine what is happening... If using a PRI type connection between the Avaya and Asterisk, it might also be useful to get some pri debug output during these tests... It could, in theory be possible to connect the Polycom directly to Asterisk, possibly with BRI, possibly through the use of H.323, but the support for both of those is somewhat less mature than much of the rest of asterisk... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP.Conf - bindaddr per peer?
31 jan 2009 kl. 02.44 skrev Mike: Replying to my own message. How difficult would it be to add a bindaddr (and possibly bindport) PER PEER in SIP.conf? How much of a bounty would I have to pay to get this done you think? Well, if you run bindaddr=0.0.0.0 Asterisk will listen to all IP's. I would say the simplest way would be to implement some sort of ACL for which address a peer accept inbound communication. The problem here is making sure that we send From the proper IP. It can be done, but with testing it's propably a couple of days work. Adding bindport would be a huge project, since it requires multiple ports in parallell, something that we're still a bit nervous about doing in chan_sip for 1.6 with the addition of TLS and TCP. The SIP structure locking scheme is... Well, to put it mildly, scary. For pricing, I would suggest you use the -biz list or send private e- mails. /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EVRC support
Hi All, I am working with asterisk 1.4 branch I need to know whether EVRC codec works with asterisk version or not? If caller and callee both has EVRC support then how the asterisk will transmit the audio with this codecs. I need to know the working role of asterisk with EVRC while it is running. Please provide information!!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
