Re: [asterisk-users] Multi-Tenant Parking
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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] Multi-Tenant Parking
Have you looked at this? http://www.google.com/#q=app_valetparking I have - but would rather use the inbuilt functionality if possible before resorting to third-party code... IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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] Multi-Tenant Parking (HALF SOLVED)
I have found that this seems to be a functional difference between the Park() and the ParkAndAnnounce() functions. Park() respects the parking lot specification, yet ParkAndAnnounce() does not respect the fact that you’ve tried to arbitrarily set the parking lot. The code below “works” as designed when the Park() function is used instead. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of UxBoD Sent: Tuesday, 12 January 2010 17:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Multi-Tenant Parking Should that not say parkinglot and not parkinglog in features.conf? It should – but that’s not a cut and paste, as the asterisk setup is on a separate, non-connected network, and I just retyped it out – not cut/paste. It’s spelt correctly in the real system (typo on here!) IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. -- _ -- 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] Multi-Tenant Parking
Has anyone managed to get multi-parking lot call parking working correctly?
I've had several attempts at it, and never seem to be able to get it to go
properly - (actually, at all):
I've most recently done this with 1.6.1.x, and now 1.6.2.x, with no luck in
either case. What I've been trying is the following:
features.conf
[general]
parkext = 100
[featuremap]
[applicationmap]
[parkinglog_customer1-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer1-park
[parkinglog_customer2-park]
parkext = 100
parkpos = 101-199
findslot = next
context = customer2-park
extensions.conf
[customer1-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into the parking
lot...)
exten = _X.,2,Set(PARKINGLOT=customer1-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer1-callback)
exten = _X.,4,Hangup()
[customer2-call-park]
exten = _X.,1,NoOp(The user ${EXTEN} is seeking place a call into the parking
lot...)
exten = _X.,2,Set(PARKINGLOT=customer2-park)
exten =
_X.,3,ParkAndAnnounce(PARKED:call-waiting,60,Local/${ext...@customer2-callback)
exten = _X.,4,Hangup()
Calls are passed to the contexts in extensions by the number of the user trying
to place the call on park. The calls park fine, can be retrieved fine, and the
callbacks work fine (via the customerX-callback) contexts which are not shown
here.
However, it simply does not seem to be putting calls into the parking lots
defined for each customer. It seems to place them all into the default parking
lot regardless of the lot you are trying to put them into. I see a lot of
people having similar issues, and I see some people claiming to have overcome
it, but no actual examples of how it was overcome.
Love anyone's input here! I'm already thinning on top - don't want to lose any
more hair on this one!
Michael Wyres
Technical Specialist
Communications Design Management
Level 1 / 99 King St
Melbourne Victoria 3000
P + 61 3 9601 6600
F + 61 3 9601 6601
mwy...@cdm.com.aublocked::mailto:sbro...@cdm.com.au
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Re: [asterisk-users] Call Limits
Would it not be easier for you to just bill them for access to 12 channels (6 extensions x 2 channels each)? Seems simpler. Then bill them for the calls they actually make. Then set call-limit=2 for each extension in sip.conf? See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Monday, 7 December 2009 00:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call Limits Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them per channel rather than per extension. A separate (but not so important) issue is that I want them to be able to make unlimited numbers of calls without their 6 extensions. So that they can have a maximum of 2 outgoing calls, and the other 4 phones can still speak to each other without having to wait for one of the 2 other calls to end. I thought that maybe one way would be to duplicate the outbound sip settings and label them outbound_client_1 and then use call-limit within that. Has anyone got any experience of this? Thanks Dan Journo Kesher Communications Ltd IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Questions about static
Is it a single user? Or every single phone? If it's a single user, and you can get hold of a UPS with power conditioning on it, try plugging the various devices into it - there might be some dirty power coming along. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Thursday, 26 November 2009 07:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Questions about static Using an Asterisk system running 1.2 with Aastra phones. Would be a cause of static for inbound/outbound and ext to ext calls? Its voip both in and out. We swapped, phones, cordes, switches etc. Typically a reboot of the phone resolves the problem...person also swears there is nothing on or near their desk to cause interference (microwave, cell phone is purse). Strange.. Thanks --Dovey IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] can't get pap2 to register from outside the LAN.
I would without the deny and permit directives in the SIP, and rule out some sort of clash there that is rejecting the address the registration is coming from, and take it from there. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Uckun Sent: Tuesday, 24 November 2009 12:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] can't get pap2 to register from outside the LAN. I am having a hell of a problem trying to get a linksys pap2t to register with my asterisk from outside the LAN. I have tried every combination of NAT, outbound proxy, stun, specify external IP address etc and it just won't work. Here are the relevant details. In asterisk I have set the following. externip=my.ip.address localnet=192.168.0.0/255.255.0.0 nat=yes bindport=5060 here is the sip user deny=0.0.0.0/0.0.0.0 type=friend secret=blah qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/372 context=from-internal canreinvite=no callgroup= callerid=device 372 accountcode= call-limit=50 I have tried nat = no, nat=never, nat=route, and leaving out the nat no difference. On the linksys end I have tried everything I can think of. Nat, no nat, stun, hard coded external IP address etc. I have read dozens of web sites and have tried every suggestion given but no joy. I know other people have had the same problem but none of the links I ran into had a solution that worked for me. This device connects perfectly when inside the lan, take it out and it won't connect no matter what I do. Here is the sip debug trace. What truly puzzles me is the 401 not authorized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. [Nov 24 14:18:41] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:41] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:42] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces - [Nov 24 14:18:42] --- (12 headers 0 lines) --- [Nov 24 14:18:42] Using latest REGISTER request as basis request [Nov 24 14:18:42] Sending to 218.101.6.157 : 5060 (NAT) [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@203.109.148.108 Content-Length: 0 [Nov 24 14:18:42] --- Transmitting (NAT) to 218.101.6.157:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108;tag=as1f31845b Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=0dc307da Content-Length: 0 [Nov 24 14:18:42] Scheduling destruction of SIP dialog 'f4e6d9bc-59a7c...@192.168.50.183' in 32000 ms (Method: REGISTER) [Nov 24 14:18:44] ip --- SIP read from 218.101.6.157:5060 --- REGISTER sip:203.109.148.108 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d From: 372 sip:3...@203.109.148.108;tag=e25fccc07a79cd65o0 To: 372 sip:3...@203.109.148.108 Call-ID: f4e6d9bc-59a7c...@192.168.50.183 CSeq: 26779 REGISTER Max-Forwards: 70 Contact: 372 sip:3...@192.168.50.183:5060;expires=3600 User-Agent: Linksys/PAP2T-5.1.6(LS) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces
Re: [asterisk-users] Setting up Nokia e71: registration problem
%20 usually represents a space in escaped URL format - perhaps you've inadvertently got a space in front of the username in the SIP account on the e71? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, 20 November 2009 13:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Setting up Nokia e71: registration problem In SIP setting on the e71 I set the public user name as 1...@10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from 'sip:%201...@10.10.11.180:5060' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[14371]: chan_sip.c:11797 check_auth: username mismatch, have %201995, digest has 1995 [2009-11-19 20:44:28] NOTICE[14371]: chan_sip.c:20481 handle_request_register: Registration from 'sip:%201...@10.10.11.180:5060' failed for '10.10.11.98' - Username/auth name mismatch so I tried changing the user name (not the public user name ) to %201995, but the e71 refused to do that since it must match the name in the public user name (Sigh). I've tried defaultuser=1995, but same result. This seems to be a problem on the e71, but is there any way asterisk can work around it. sean ___ -- 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 IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] can't call through voip provider
To be perfectly complete, exactly which inbound ports to open will depend on the phones in use. For example, a Cisco 7940 (using this example because I have one on my desk at the moment), the default ports from the config are: voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 Meaning, you have to have 5060 open (obviously), and all the ports between the start and end media port. Many phones will let you adjust where these boundaries lie, but some won't. You'll need enough range to cover every kind of phone (soft or hard) that you are using. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Thursday, 19 November 2009 09:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider Ok. I do NOT have ports 1-2 opened in. I guess I should try that and see if it works. I will open ports 5060 - 5070 and 1 - 100100 and do some test tonight. I will keep you posted. Thanks. --- On Wed, 11/18/09, Danny Nicholas da...@debsinc.com wrote: From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] can't call through voip provider To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:18 PM According to what I know, you have to have 5060 open out and 1-2 open in (you can cut this to as small as 1-10004). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Wednesday, November 18, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't call through voip provider According to the provider he says he doesn't see anything coming in on their side. I've had all ports FORWARD out to ACCEPT but, blocking incoming new connections. I thought when asterisk starts a communication with a remote server using an unprivate port to port 5060 theres already an ESTABLISHED communication. I don't know if I'm having problems with my firewall script or what but, since there isn't any new connections coming form outside I think I'm ok to accept only ESTABLISHED,RELATED coming in. I don't know but, I'm stuck with this problem and don't know what else to do. --- On Wed, 11/18/09, Warren Selby wcse...@selbytech.com wrote: From: Warren Selby wcse...@selbytech.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 5:03 PM What does your provider see when you attempt to call them? Thanks, --Warren Selby On Nov 18, 2009, at 3:38 PM, Landy Landy landysacco...@yahoo.com wrote: Thanks for replying. But how come I'm able to use a softphone to place calls from withing the lan? I really dont get it. What ports should I enable in the INPUT chain? --- On Wed, 11/18/09, Jared Smith jsm...@digium.com wrote: From: Jared Smith jsm...@digium.com Subject: Re: [asterisk-users] can't call through voip provider To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, November 18, 2009, 9:28 AM On Wed, 2009-11-18 at 06:01 -0800, Landy Landy wrote: Please help me with this, I can find any solution on this pls help. Your help will be very appreciated. Thanks. It appears that Asterisk keeps sending an SIP INVITE message to your provider, but not getting any kind of response. After a number of attempts at re-transmitting the message, it's giving up. You need to check your network configuration and find out why responses from the provider aren't getting back to your Asterisk system. This is typically a problem with firewalls, either on the Asterisk system itself or between Asterisk and your VoIP provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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
Re: [asterisk-users] Queues
Hi Travis, There's lots of different ways to attack on-call roster solutions in Asterisk - as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn't always suit the business need. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his on-call roster routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list - they could simply select the right person, and click a First Call button to make that person the first in the roster, select another person and click a Second Call button to make that person the second in the roster, and so on. Using the Asterisk manager interface - (or even asterisk -rx command if you're not comfortable using the AMI) - you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk - there is more than one way to hump the leg. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is extension-based, you have at least two options. Option 1 is to have a few extensions designated for following where you punch in the cell numbers as you wish. Option 2 is to use day logic to point to the following guys based on days.If I were doing option 2, I'd try to use ASTDB to control this instead of having to code a lot of dialplan, but that's just me... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queues Hello Everyone, I'm looking for help/ideas on how to do the following: I have a couple of people out of many (the couple of people randomly change) who log into an on-call queue. A call comes in and it rings the on-call extensions, but no one answers. I would like the call to then try the cell-phones of just the people that are logged into the on-call queue. I've got the queue setup and the people log into and out of it by dialing extensions that use AddQueueMember() and RemoveQueueMember() respectively. I tried using QUEUE_MEMBER_LIST to write to a database list when the call comes in however it keeps adding duplicates each time the call goes into the queue. I'm just not seeing how to pass the call that goes into the queue to a dynamic list on the way out. Is attempting something like this even realistic? Thanks in advance, Travis ___ -- 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 IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named
Re: [asterisk-users] Queues
Again – lots of ways to do it – you could use a web interface to set the numbers in ASTDB for lookup – or you could create an IVR to ask for the number, and store it in ASTDB that way. Good luck! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Tuesday, 17 November 2009 10:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues Hi Michael, Your web interface for the on-call roster is pretty close to what we're trying to trying to achieve. I would like to have people signing into the on-call queue be the method that determined whose cell phone to call. I was hoping there was a way to pass the call exiting the queue to a variable or two that was composed of the extensions currently logged into the queue. I set up an extension number for people to call into and enter a forwarding number which writes an entry into the ASTDB. I have my dialplan check to see if there is an ASTDB entry for that extension before it tries to dial their deskphone, and if there is an entry it dials the forwarded number stored in the database instead. The closest thing so far I have found to what I am trying to achieve is to hard code a couple of spare extensions into the dialplan, and then have whoever is on-call set one of those extensions to their cell phone number. I'll definitely take another look at followme to see if I can adapt that what I'm trying to achieve. Thanks Danny Michael, Travis - Original Message - From: Michael Wyres mwy...@cdm.com.au To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 2:23:49 PM Subject: Re: [asterisk-users] Queues Hi Travis, There’s lots of different ways to attack “on-call” roster solutions in Asterisk – as Danny suggested, FollowMe() is definitely an option (and normally the best), but it doesn’t always suit the “business need”. However, also as Danny suggested, in most cases using ASTDB in some way to simplify dialling plans is the way to go - then you just have to decide how you want to update the information as to the number to call, in ASTDB. For example, I had a customer a couple of years back who desperately wanted to manage his “on-call roster” routing using a web interface. I dollied up a simple PHP/MySQL web interface with a list of all the people (and their mobile/cell numbers) in a drop down list – they could simply select the right person, and click a “First Call” button to make that person the first in the roster, select another person and click a “Second Call” button to make that person the second in the roster, and so on. Using the Asterisk manager interface – (or even “asterisk –rx command” if you’re not comfortable using the AMI) – you get the numbers selected into ASTDB. The dialplan just comes along then and reads the appropriate numbers from ASTDB as it steps through, and dials the people in order. As with many things in Asterisk – there is more than one way to “hump the leg”. Cheers Michael From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, 17 November 2009 08:57 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queues Since followme is “extension-based”, you have at least two options. Option 1 is to have a few extensions designated for “following” where you punch in the cell numbers as you wish. Option 2 is to use “day logic” to point to the “following” guys based on days.If I were doing option 2, I’d try to use ASTDB to control this instead of having to code a lot of dialplan, but that’s just me… From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queues I had looked at followme as a solution but ran into the same stumbling block of having to hard code the cell phone list. I didn't see a dynamic way of the list being extensions 12 and 14 on Monday, but changing to extensions 13 and 19 on Tuesday without editing the extensions.conf file manually each day. Did I overlook something in how followme works? - Original Message - From: Danny Nicholas da...@debsinc.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 16, 2009 1:37:04 PM Subject: Re: [asterisk-users] Queues It should be realistic, but have you considered just using followme to add the cell phones to the queue list? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Travis Elsberry Sent: Monday, November 16, 2009 3
Re: [asterisk-users] Changing labels on Phones
Sometimes they reboot when you try this, but usually not - but you can just change one setting in the network configuration (eg: change the phones IP address), and it will go through just the very last part of it's normal boot process, and re-pull it's TFTP configuration, and update things - still takes around 20 to 30 seconds, so not really what you are looking for - but I've been looking for a way to do this for about three and a half years! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: Monday, 16 November 2009 08:51 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Changing labels on Phones We have several types of phones, Cisco 79xx, Aastra 9133i etc. We have a hotdesk type system where anyone can log on to an extension - however what I would love to do is relabel the phone with the current owner when this logon happens. I know that I can change the sip.conf and phones tftp file, however this is a big problem with the Cisco's as they take *forever* (ok, maybe 2 / 3 minutes) to reboot (VLAN problem) 1) Has anyone actually solved this VLAN issue with the cisco ? 2) Is there any way of changing a label without rebooting the phone ? TIA Julian IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Hardware Requirement for asterisk
Throwing him off the list would not achieve anything - he still has our email addresses, and will still be able to send you email. Unless of course, you pop his email address on the DENY list of your gateway...*whistles innocently* From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andreas Anderson Sent: Monday, 16 November 2009 06:34 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Hardware Requirement for asterisk i am going to set up asterisk for pbx purpose in my office. i am having 2 PSTN lines and will be configuring 10 extentions in my office. plz tell me which hardware will be needed for this. Can someone please throw that moron of the list?? Windows 7: Make your own home movies. Learn more.http://download.live.com/moviemaker IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] allowguest defaults to yes for SIP
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee Howard Sent: Friday, 13 November 2009 06:16 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] allowguest defaults to yes for SIP I could be wrong, but I don't generally consider myself stupid or lazy... and yet this default setting as yes took me by surprise, obviously. This has nothing to do with stupidity or laziness. The way I see it, the reason you have encountered some resistance to your opinion in regards to whether guest access should be allowed by default or should not be, is not because your opinion is right or wrong - everyone is entitled to an opinion - and your stance has merit, certainly - I don't think anyone is actually disputing that. It is more that a lot of the people on this list have been using Asterisk for a LNG time, and have explained why it might be advantageous to have guest access enabled by default. There are definitely uses for this functionality, as has been demonstrated by a number of examples contained in this thread. Isn't this why you joined the list? To learn more about the product, and get ideas and assistance from the more experienced users of the product? You raised your concern, and Tilghman (a senior developer at Digium) explained the reasoning behind the default setting. He suggested that you take your concern to the tracker and post a patch. You resisted. The open source community (despite what some think) is a highly organised community, with structures in place to get things like that done. If you consistently did end runs around established corporate procedures in your workplace, you'd expect a foot up the ass from management. Tilghman was as politely as possible asking you to follow the established procedures. You chose to resist. Now, the default extensions.conf contains the following snippet: snip [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include = demo /snip Now, a lot of people never RTFM for anything. Moreover, how many people actually read the EULA for any piece of software they use? It's not Asterisk/Digium's fault if people don't read the available documentation that they provide. The quite plainly clear statement above is in a production system, you probably don't want to have the demo there. Did you read that bit? Did you wonder why that bit is there? When I first started working with Asterisk, I clearly remember that line (or something very similar) piquing my curiousity to dig a little deeper as to why that statement was made. Lo, I discovered that this was because by default, guest access is allowed. Digium has made that available in the distribution for EVERYONE to read, and extensions.conf is probably the most accessed file in an Asterisk system not using RealTime, so people who choose to ignore reading the excellent notes and annotations in all of the default configuration files is doing themselves a disservice. I too found the default access odd at first, but I chose to understand the reasoning from people who knew better, instead of chucking a hissy fit. IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Bad quality of call
The reasons for poor call quality are many and varied. As another poster suggested, the headset you are using might be poorly configured, or just a poor example. An under-spec server could also do it - I use two simple, low-spec Virtual Machines in my dev lab that I bring up when I want to test dialplan variations, without interfering with the live systems. Quality is usually terrible because they are underpowered, but I'm only quickly testing dialplans, so I'm not concerned about the quality in those instances. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo lombardo Sent: Thursday, 12 November 2009 01:02 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bad quality of call Hi all, I did some call using an asterisk 1.4 PBX and 2 softphone in a private network; call is up, but with bad quality. Someone knows how to debug this problem ? Thanks in advance for any help. -- Giancarlo Lombardo IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Can't connect to voip provider over NAT
Have you tried nat=yes in the definition in sip.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Landy Landy Sent: Thursday, 12 November 2009 13:30 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't connect to voip provider over NAT Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=theprovider's server username=username secret=password port=5060 canreinvite=YES dtmfmode=rfc2833 I've tried opening all ports to test this but, still doesn't work. Now, I need to know which especific ports to open in order to allow sip flow correctly. Also enabled/opened ports 5060 - 5070 and the rtp: rtpstart=1 rtpend=2 Don't know what else to try. Please help. Thanks in advanced for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users IMPORTANT NOTICE TO RECIPIENT Computer viruses - It is your responsibility to scan this email and any attachments for viruses and defects and rely on those scans as Communications Design Management Pty Limited (CDM) does not accept any liability for loss or damage arising from receipt or use of this email or any attachments. Confidentiality - This email and any attachments are intended for the named recipient only and may contain personal information, be it confidential or subject to privilege, none of which are lost or waived because this email may have been sent to you in error. If you are not the named addressee please let CDM know by return email, permanently delete it from your system and destroy all copies and do not use or disclose the contents. Copyright - This email is subject to copyright and no part of it maybe reproduced in any manner without the written permission of the copyright owner. Privacy - Within the jurisdiction of Australian law, personal information in this email must be dealt with in compliance with the Australian Federal Privacy Act 1988. ___ -- 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] Call declined
Try: [tutorial] exten = 1234,1,Dial(SIP/gianca,10,t) exten = 12345,1,Dial(SIP/giusy,10,t) You want a / between SIP and the name of the phone, not an ,. The 10 refers to the number of seconds you want the phone to ring. The t allows the channel to be transferred after pickup - not strictly needed, but I tend to put it in in most instances as generally you'll want it. For more information on the Dial application, see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo lombardo Sent: Tuesday, 10 November 2009 09:03 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call declined Dear all, I'm in basic setup of my network: I try to do a call from a softphone to an other one but I got the error 603 Declined. Below the sip.conf: [gianca] type=friend username=gianca secret=pwd_gianca host=dynamic context=tutorial [giusy] type=friend username=giusy secret=pwd_giusy host=dynamic context=tutorial extension.conf: [tutorial] exten = 1234,1,Dial(SIP,gianca) exten = 12345,1,Dial(SIP,giusy) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862http://sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265 v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (12 headers 11 lines) --- Sending to 192.168.1.116 : 14862 (NAT) Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. --- Reliably Transmitting (no NAT) to 192.168.1.116:14862http://192.168.1.116:14862 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100;tag=as29d2b71c Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY upported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=42ebb35e Content-Length: 0 Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE) Found user 'gianca' dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- ACK sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100;tag=as29d2b71c From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 1 ACK Content-Length: 0 - --- (7 headers 0 lines) --- dhcppc0*CLI --- SIP read from 192.168.1.116:14862http://192.168.1.116:14862 --- INVITE sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport Max-Forwards: 70 Contact: sip:gia...@192.168.1.116:14862http://sip:gia...@192.168.1.116:14862 To: 12345sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100 From: giancasip:gia...@192.168.1.100mailto:sip%3agia...@192.168.1.100;tag=db428348 Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username=gianca,realm=asterisk,nonce=42ebb35e,uri=sip:12...@192.168.1.100mailto:sip%3a12...@192.168.1.100,response=8d00b3e1b28ed2e40681a3a9ee410046,algorithm=MD5 User-Agent: X-Lite release 1103k stamp 53621 Content-Length: 265 v=0 o=- 6 2 IN IP4 192.168.1.116 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.116 t=0 0 m=audio 5960 RTP/AVP 107 0 8 101 a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 192.168.1.116 :
