Re: [asterisk-users] Latency woes, qos the fix?

2008-10-18 Thread Alex Balashov
Stephen Reese wrote: >> 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 hos

Re: [asterisk-users] Latency woes, qos the fix?

2008-10-18 Thread Stephen Reese
> 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 traffi

Re: [asterisk-users] Latency woes, qos the fix?

2008-10-18 Thread Alex Balashov
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

Re: [asterisk-users] IP Address on CDR

2008-10-18 Thread Alex Balashov
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

Re: [asterisk-users] IP Address on CDR

2008-10-18 Thread Juan Rodríguez
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 > > ___ > -- Bandwidth and Colocat

[asterisk-users] Latency woes, qos the fix?

2008-10-18 Thread Stephen Reese
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

[asterisk-users] app_confcall on Asterisk 1.6 update

2008-10-18 Thread jonathan augenstine
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 htt

[asterisk-users] IP Address on CDR

2008-10-18 Thread Nhadie
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.

Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

2008-10-18 Thread Stephen Reese
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

[asterisk-users] Does asterisk 1.6 support an authname with a domain?

2008-10-18 Thread Eric Chamberlain
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 Cham

[asterisk-users] Is there a way to specify the fromdomain from the dialplan?

2008-10-18 Thread Eric Chamberlain
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

Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

2008-10-18 Thread Darryl Dunkin
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 Di

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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, b

Re: [asterisk-users] Cisco 7960 not always receiving incoming calls

2008-10-18 Thread Stephen Reese
> 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 dev

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Steve Totaro
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

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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 potenti

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Steve Totaro
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" i

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Joseph
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

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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 poin

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Joseph
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

Re: [asterisk-users] OT: Polycom IP330 user problem

2008-10-18 Thread Doug Lytle
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'

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Joseph
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 fire

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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 yo

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Joseph
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

[asterisk-users] OT: Polycom IP330 user problem

2008-10-18 Thread Bill Michaelson
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

[asterisk-users] strange h323 delay issue

2008-10-18 Thread Giedrius Augys
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

[asterisk-users] Looking to replicate OnSIP ........SER + Asterisk

2008-10-18 Thread EdPimentl
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 ] __

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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 u

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
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.

Re: [asterisk-users] SER + Asterisk

2008-10-18 Thread Alex Balashov
No, a proxy cannot *initiate* anything. ram wrote: > > > On Sat, Oct 18, 2008 at 9:20 AM, Joseph <[EMAIL PROTECTED] > > wrote: > > On 10/17/08 23:23, Kristian Kielhofner wrote: > >On 10/17/08, Alex Balashov <[EMAIL PROTECTED] > >

Re: [asterisk-users] Asterisk 1.4 and openLDAP

2008-10-18 Thread Tilghman Lesher
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 w

Re: [asterisk-users] Asterisk 1.4 and openLDAP

2008-10-18 Thread Ming Yong
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=Xl78orka

[asterisk-users] Asterisk 1.4 and openLDAP

2008-10-18 Thread Anael DIAZ
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-u