Re: [asterisk-users] SER + Asterisk
On 10/18/08 05:28, Grey Man wrote: As far as I'm aware SER (and it's derivatives) cannot initiate outbound registraitions. They can do the opposite and act as a SIP Registrar. For outbound registrations you should be able to use Asterisk. Regards, Greyman. Yes, I use Asterisk for iax outside registration but not for sip from Asterisk; I don't want to make a swizz cheese (open so many ports) out of my firewall. I can not use stun with Asterisk. I have my stand alone sip phone registered with the provider but I can only register it with one provider so no Asterisk access. What are my best options? I was thinking that something like nathelper with SER would be of any use to me but I see it might not be the case, it is only for helping to register clients IN not OUT. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
On 10/18/08 10:39, ram wrote: On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] wrote: I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] Asterisk 1.4 and openLDAP
Hi there, I need help in implementing Asterisk with LDAP. I' ve installed Asterik 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD. 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
Re: [asterisk-users] Asterisk 1.4 and openLDAP
Anael, You should take a look at Druid (Open Source Unified Communications) Project based on Asterisk that has complete LDAP backend and Zimbra connector. It's an open source project we are looking for collaborators users. Druid UCS 5.0 with LDAP backend http://www.youtube.com/watch?v=Xl78orka938 Druid Zimlet for click to call and drag drop faxing http://www.youtube.com/watch?v=WdEVSJuh1ow Ming On Sat, Oct 18, 2008 at 3:30 PM, Anael DIAZ [EMAIL PROTECTED] wrote: Hi there, I need help in implementing Asterisk with LDAP. I' ve installed Asterik 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD. 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 -- Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com Voiceroute videos on Druid, Open Source Unified Communications Asterisk http://youtube.com/voiceroute ___ -- 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 1.4 and openLDAP
On Saturday 18 October 2008 02:30:16 Anael DIAZ wrote: I need help in implementing Asterisk with LDAP. I' ve installed Asterik 1.4 with CentOS 5.2 and I would like to use with it an existing zimbra LPAD. You might want to take a look at Asterisk 1.6, which has LDAP realtime support. Look within contrib/scripts to find a working example of an LDIF and schema file for use with Asterisk. -- 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] SER + Asterisk
No, a proxy cannot *initiate* anything. ram wrote: On Sat, Oct 18, 2008 at 9:20 AM, Joseph [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: On 10/17/08 23:23, Kristian Kielhofner wrote: On 10/17/08, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. Slight clarification: Kamailio (formerly OpenSER) / OpenSIPS is the thing to do. I would gladly go with any of the newer packages if I only could. I'm just working with what I can find in portage; I'm sure it will be eventually available. It will first show up via overlay. What I'm trying to do is to register SER to my VoIP provider via stun.fwdnet.net http://stun.fwdnet.net/ and connect SER with Asterisk, I just need some simple practical example; and upgrade will come with time. I'm sure it is possible even with old SER. Suggesting what is newer is not going to help me much :-) Hi Joseph you can use UAC Module to register with provider and make calls using SER/Openser/OpensSIPs or you can do other way is SER as registrar and Asterisk act a b2bua ( you can register with provider) let me know if it helps your need Ram ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Doesn't mean it's not defunct. Joseph wrote: I'm using Gentoo and the only package I was able to find in portage was SER; I could compile manually but it is harder to upgrade and keep track of dependencies. -- #Joseph On 10/17/08 22:42, Alex Balashov wrote: SER is defunct. Kamailio / OpenSIPS (formerly OpenSER) is the thing to do. On Fri, October 17, 2008 9:36 pm, Joseph wrote: I am running Asterisk and would like to add SER to register my (sip) DID and connect it to asterisk; but I'm not sure if this is the correct forum. I have as DID, sip account with one VoIP provider; currently Im using just stand alone SIP phone and register with the VoIP provider via: stun.fwdnet.net Is it possible to use SER to register with the provider and forward the call Asterisk. Can anybody provide a link to practical example. I'm comfortable with Asterisk but I just install SER and can not find appropriate example to follow on www.iptel.org web-page. There are a lot explanations but not enough practical examples to follow. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. You cannot use the UAC module to register. The proxy is an event-driven element, by definition; it cannot initiate anything, nor can it itself possess UAC credentials. What you can do with the UAC module is take advantage of the proxy's ability to statelessly or statefully forward calls, branch calls, and reply to particular feedback by mimicking some of the behaviour of a UAC and/or sending an authentication digest in response to a registration or proxy challenge. But you can't use it to register with a provider as such. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Looking to replicate OnSIP ........SER + Asterisk
Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] ___ -- 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 h323 delay issue
Hello, I have a strange h323 issue. After executing command Dial(SIP/333-0d1dfe00, H323/[EMAIL PROTECTED]|5|tT) at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what asterisk (h323 channel) was doing for 40 sec??? What reasons could invoke this problem? I haven't any problems with SIP channels. My versions: asterisk-1.4.21.1 asterisk-addons-1.4.6 openh323_v1_18_0 pwlib_v1_10_0 My h323.conf configurations: [general] port = 1720 bindaddr = 192.168.1.165 tos=lowdelay disallow=all allow=g729 dtmfmode=rfc2833 gatekeeper = DISABLE AllowGKRouted = no AcceptAnonymous = no context=from-trunk [ccg] type=friend context=from-trunk host=192.168.1.163 port=1720 disallow=all ;allow=alaw ;allow=ulaw allow=g729 fastStart=yes h245Tunneling=yes A full log: [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/333-0d1d8fb0, H323/[EMAIL PROTECTED]|5|tT) in new stack [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: type=H323, format=8, [EMAIL PROTECTED] [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Extension: 361737052390920 Host: ccg [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Calling to [EMAIL PROTECTED] H323/ccg-2 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Requested transfer capability: 0x00 - SPEECH [Oct 18 22:32:23] DEBUG[18236] chan_h323.c: Placing outgoing call to [EMAIL PROTECTED]:1720, 101 [Oct 18 22:32:23] VERBOSE[18236] logger.c: -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. [Oct 18 22:32:23] VERBOSE[18236] logger.c: Using 192.168.1.165 for outbound call [Oct 18 22:33:03] VERBOSE[18236] logger.c: == New H.323 Connection created. [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- root is calling host [EMAIL PROTECTED]:1720 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call token is ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Call reference is 6453 [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- DTMF Payload is [pt=101] [Oct 18 22:33:03] VERBOSE[18236] logger.c: -- Called [EMAIL PROTECTED] [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting capabilities for connection ip$localhost/6453 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Setting capabilities to 0x100 (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Capabilities in preference order is (g729) [Oct 18 22:33:03] VERBOSE[18238] logger.c: Allowed Codecs: [Oct 18 22:33:03] VERBOSE[18238] logger.c: Table: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: Set: [Oct 18 22:33:03] VERBOSE[18238] logger.c:0: [Oct 18 22:33:03] VERBOSE[18238] logger.c: 0: [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729A 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c:G.729 2 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 1: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/hookflash 3 [Oct 18 22:33:03] VERBOSE[18238] logger.c: 2: [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/RFC2833 4 [Oct 18 22:33:03] VERBOSE[18238] logger.c:UserInput/dtmf 5 [Oct 18 22:33:03] VERBOSE[18238] logger.c: [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Capabilities for connection ip$localhost/6453 is set [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Created RTP channel [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting NAT on RTP to 0 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Sending RTP 'US' 192.168.1.165:23786 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Sending SETUP message [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Transmitting RFC2833 on payload 101 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- Started logical channel: sending G.729A [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- channelsOpen = 1 [Oct 18 22:33:03] VERBOSE[18238] logger.c: External RTP Session Starting [Oct 18 22:33:03] VERBOSE[18238] logger.c: RTP channel id 1 parameters: [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- remoteIpAddress: 192.168.1.163 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- remotePort: 10626 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- ExternalIpAddress: 192.168.1.165 [Oct 18 22:33:03] VERBOSE[18238] logger.c: -- ExternalPort: 23786 [Oct 18 22:33:03] DEBUG[18238] chan_h323.c: Setting up RTP connection for ip$localhost/6453 [Oct 18
[asterisk-users] OT: Polycom IP330 user problem
I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. --- I looked at the server logs for the phone call missed by . They indicate that the call came in at 15:32:25, and was routed to her telephone at 15:32:32. This timed out after about 25 seconds as it should if unanswered, and was sent to voicemail at 15:32:58. I called BB and asked her to check the phone display. She told me that the phone logged an unanswered call at 15:32:32, precisely in accordance with the server log. This leaves two possible conjectures: * The telephone, for whatever reason, did not ring in response to the incoming call signal which it obviously received. * The telephone ringer was not audible or noticeable to for some other reason. For the first possibility, I can think of three circumstances that would cause this: * If the handset is slightly ajar, i.e., off-hook, the phone will make no sound, but log the call. Upon receipt of the message waiting notification, it will start blinking. Eventually, the phone reverts to on-hook status by itself even if the handset is still ajar. * If the alert code for silent ring is set, the line annunciator will flash silently to indicate the call coming in. * If the phone is malfunctioning anything can happen. There is no indication that silent ring alert was set, nor is there any current configuration setting that should cause this. That leaves three bullet points for us to consider. I can follow up with one: I will research this as thoroughly as I can to see if there are any reports of malfunctions by Polycom IP330 phones that conform to this behavior, or if there are any other possible explanations for the events that I've overlooked. If you would like to follow up in any other way, let me know what I can do to help. smime.p7s Description: S/MIME Cryptographic Signature ___ -- 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] SER + Asterisk
On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco -- #Joseph ___ -- 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] SER + Asterisk
Joseph wrote: On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco Well, you *can* use the proxy to provide near-end NAT traversal. The UAC module won't help much here; your best bet is to statefully relay the REGISTER messages and the corresponding challenges. There is a nathelper module that can help you fix up the contact bindings if it they contain RFC1918 addresses. However, it should be emphasised in no uncertain terms that your UAC (Asterisk) must originate the request and relay it through the proxy; the proxy cannot originate it itself. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 15:31, Alex Balashov wrote: Joseph wrote: On 10/18/08 13:51, Alex Balashov wrote: Joseph wrote: Thanks for your help. How to use UAC Module to register with a provider? Is there something like STUN for SER? I don't want to open too many ports on my firewall. You do not need to open any ports on your firewall if your NAT gateway does proper translation. No, my firewall does not support NAT gateway translation, it is freesco Well, you *can* use the proxy to provide near-end NAT traversal. The UAC module won't help much here; your best bet is to statefully relay the REGISTER messages and the corresponding challenges. There is a nathelper module that can help you fix up the contact bindings if it they contain RFC1918 addresses. However, it should be emphasised in no uncertain terms that your UAC (Asterisk) must originate the request and relay it through the proxy; the proxy cannot originate it itself. Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] OT: Polycom IP330 user problem
Bill Michaelson wrote: I recently sent this email to a user in response to a problem report of phone calls going to voicemail without the phone ringing. I'm wondering if I've covered all bases, or whether there is some logical explanation I haven't considered, and generally what others' opinions/experiences are that relate. This is an Asterisk system, of course. Have you actually witnessed this behavior yourself? I do have people that like to ignore calls that they'll swear that the phone never rang. BUT, it always seems to work just fine when I'm around. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SER + Asterisk
On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
Joseph wrote: On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On 10/18/08 16:48, Alex Balashov wrote: [snip] There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. I totally agree with you, it is hard to understand, I've been telling all alone that programmers shouldn't write manuals :-) I'm in a stage that even if I stole a recipe from someone I wouldn't know what to do with it :-/ -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 4:48 PM, Alex Balashov [EMAIL PROTECTED]wrote: Joseph wrote: On 10/18/08 16:10, Alex Balashov wrote: Joseph wrote: Thanks for the info Alex, Do you have a good links that would help accomplish it? I was under impression that nathelper is only for incoming connection, not outgoing. Sure - it's incoming from the point of view the proxy, if you do: Asterisk --- proxy w/NAT traversal fixups --- provider :-) Any links I can think of that explain how to use nathelper rely on a pre-existing knowledge of how to deal with OpenSER, which is a rather esoteric and low-level topic compared to Asterisk. I'm trying to find a good manual for SER with decent examples for beginners but don't have much luck. I think these package OpenSER OpenSIPS SER are not so common as it is hard to understand them. The manual on their web-page is just dry plain language without examples so it makes it harder to understand. There is not really a lot of good conceptual introduction to OpenSER, although Flavio Goncalves' book (Building Scalable Telephony Applications With OpenSER) may be somewhat of aid. The documentation primarily serves those that already know what they are doing, kind of like programmers that just need an API reference. But basically, it is admittedly a lot of work to figure out an extremely polymorphic and idiosyncratic environment just to solve a relatively simple problem. I recommend contracting someone to take care of it for you, or stealing a recipe from somewhere. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 Jeremy McNamara wrote some helpful tidbits as well. http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/ This one is a Configuration Wizard I haven't tried it out yet but certainly will at some point http://www.jeremy-mcnamara.com/2007/02/22/seropenser-configuration-wizard/ If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] SER + Asterisk
Steve Totaro wrote: If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! The problem with simply doing a GUI frontend to *SER is that it's very polymorphic far too extensible; there are far too many potential applications, and those applications are far too customised and situation-specific. That's why the routing script takes the character that it does, because it wishes to have as few cookie-cutter characteristics as possible. That having been said, there are plenty of common use cases of the product which probably deserve GUI implementation. But it needs to be understood that they are just common use cases, nothing more, and represent an infinitesimal fraction of conceivable -- and routine -- applications. The product is far too low-level to be able to say what it does even in the loose ways in which we routinely attribute certain functional goals or traits to Asterisk. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] SER + Asterisk
On Sat, Oct 18, 2008 at 5:35 PM, Alex Balashov [EMAIL PROTECTED]wrote: Steve Totaro wrote: If someone wrote a nice webmin module with all the configuration options as check boxes and fill in the blanks, that would be very NICE! The problem with simply doing a GUI frontend to *SER is that it's very polymorphic far too extensible; there are far too many potential applications, and those applications are far too customised and situation-specific. That's why the routing script takes the character that it does, because it wishes to have as few cookie-cutter characteristics as possible. That having been said, there are plenty of common use cases of the product which probably deserve GUI implementation. But it needs to be understood that they are just common use cases, nothing more, and represent an infinitesimal fraction of conceivable -- and routine -- applications. The product is far too low-level to be able to say what it does even in the loose ways in which we routinely attribute certain functional goals or traits to Asterisk. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 Kind of like SwitchVox, FreePBX, Thirdlane.. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Cisco 7960 not always receiving incoming calls
As a last resort (if qualify doesn't help), you could enter this (global) to increase the timeout on UDP translations: ip nat translation udp-timeout 300 (or greater if you prefer) It is likely a NAT timeout issue. When you call outbound, you 'reactivate' the SIP session in your NAT device, allowing calls to come in until it expires (default on many devices is 60 seconds). You may also receive inbound calls when the phone reregisters regularly. Try 'qualify=yes' in your phones section in sip.conf to send keepalives (option packets in this case) every two seconds to the phone to keep it from going idle. You can see the state of the phone from the console with a 'sip show peers', if unreachable, your NAT device has killed the NAT forward. Should look like one of these: xxx/xxx x.x.x.x D N 5060 OK (46 ms) xxx/xxx x.x.x.x D N 5060 UNREACHABLE As another troubleshooting step, you can telnet to the phone and have it reregister with Asterisk manually (register line 1 1) to see if that brings it back to life. If qualify doesn't do it, see if you can increase UDP timeouts in your firewall/NAT device. I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki
Re: [asterisk-users] SER + Asterisk
Steve Totaro wrote: Kind of like SwitchVox, FreePBX, Thirdlane.. I don't know that I'd make that comparison. I would say that in general, OpenSER is more low-level and amorphous and multipurpose than Asterisk or any GUI that wraps it. Asterisk has many applications and uses and niches, but these are all uses that capitalise on the sort of thing that Asterisk is. The genus of thing that it is on a technical level and the role it plays is fairly well-understood, even if there are many things you can do with that particular type of thing. OpenSER is hard to pin down like that. The closest you can come to it is to say that it is a proxy/UAS, and what does that really get you? It's used in situations that offer far less taxonomic resemblance to each other than sundry appropriations of Asterisk do. Yes, there's no argument that there are many things OpenSER does that can be driven by a GUI. But at the same time, that approach is somewhat antithetical to its basic nature. Its roles cannot be usefully anticipated nearly as well or as much. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Cisco 7960 not always receiving incoming calls
Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0 crypto isakmp profile IKE-PROFILE match identity group VPN-Users client authentication list default isakmp authorization list default client configuration address initiate client configuration address respond virtual-template 1 ! ! crypto ipsec transform-set ESP-3DES-SHA esp-3des
[asterisk-users] Is there a way to specify the fromdomain from the dialplan?
Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)[EMAIL PROTECTED] The SIP From header turns into: [EMAIL PROTECTED]@10.10.10.10 We want [EMAIL PROTECTED], and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain ___ -- 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] Does asterisk 1.6 support an authname with a domain?
We need to include the domain information in the Authentication digest username SIP header field. Using SIP/username[:password[:md5secret[:[EMAIL PROTECTED]:port] in the dialplan breaks if authname needs [EMAIL PROTECTED], is there a way to specify this value from the dialplan? -- Eric Chamberlain ___ -- 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] Cisco 7960 not always receiving incoming calls
Very cool, I believe that did the trick. Thank you for your time. On Sat, Oct 18, 2008 at 7:42 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0
[asterisk-users] IP Address on CDR
Hi, How can i log the IP address of the caller on asterisk mysql cdr? Regards, Nhadie ___ -- 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] app_confcall on Asterisk 1.6 update
FYI I was informed by A. Minnesale that app_confcall was originally developed for Asterisk 1.2. He stated that there would probably be a significant amount of work to update it to Asterisk 1.6. Jonathan ___ -- 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] Latency woes, qos the fix?
My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.111D N 1038 OK (133 ms) This seems pretty high when my ping time from a host on the same network is ~30ms: Pinging 209.251.157.93 with 32 bytes of data: Reply from 209.251.157.93: bytes=32 time=30ms TTL=51 Reply from 209.251.157.93: bytes=32 time=27ms TTL=51 Reply from 209.251.157.93: bytes=32 time=36ms TTL=51 Reply from 209.251.157.93: bytes=32 time=28ms TTL=51 Any suggestions or is this normal? Should I enable qos on my Cisco 3725 router and 2950 switch? Would I also need to enable the following in the sip.conf ;tos_sip=cs3; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5; Sets 802.1p priority for RTP audio packets. ;cos_video=4; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets ___ -- 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] IP Address on CDR
Maybe you can use ${SIP_HEADER(FROM)}.
Regards,
Juan
On Sat, Oct 18, 2008 at 10:31 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
How can i log the IP address of the caller on asterisk mysql cdr?
Regards,
Nhadie
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Juan E. RodrÃguez
Cel. 829-886-5565
Work: 809-724-9227
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Re: [asterisk-users] IP Address on CDR
Define IP address of the caller? From header, Contact, literal IP source of request...? Nhadie wrote: Hi, How can i log the IP address of the caller on asterisk mysql cdr? Regards, Nhadie ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Latency woes, qos the fix?
Does the latency remain more or less the same regardless of the bandwidth load on the pipe? If so, TOS bits (what you refer to as QoS) won't help you. You've either got network issues (very likely if you have an intra-network ping of 30 ms) or the outside host you're sending the traffic to is just that far away in latency terms. Stephen Reese wrote: My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.111D N 1038 OK (133 ms) This seems pretty high when my ping time from a host on the same network is ~30ms: Pinging 209.251.157.93 with 32 bytes of data: Reply from 209.251.157.93: bytes=32 time=30ms TTL=51 Reply from 209.251.157.93: bytes=32 time=27ms TTL=51 Reply from 209.251.157.93: bytes=32 time=36ms TTL=51 Reply from 209.251.157.93: bytes=32 time=28ms TTL=51 Any suggestions or is this normal? Should I enable qos on my Cisco 3725 router and 2950 switch? Would I also need to enable the following in the sip.conf ;tos_sip=cs3; Sets TOS for SIP packets. ;tos_audio=ef ; Sets TOS for RTP audio packets. ;tos_video=af41 ; Sets TOS for RTP video packets. ;tos_text=af41 ; Sets TOS for RTP text packets. ;cos_sip=3 ; Sets 802.1p priority for SIP packets. ;cos_audio=5; Sets 802.1p priority for RTP audio packets. ;cos_video=4; Sets 802.1p priority for RTP video packets. ;cos_text=3 ; Sets 802.1p priority for RTP text packets ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
