similar to: qualify and NAT....

Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway,

Hi Everyone, I'm new to asterisk and trying to get together the hardware to run a few POTS phone extentions and one or two POTS lines for starters. For these low port counts, I could just go with FXS and FXO cards, but... I can get a Cisco MC3810 with a mixture of FXO and FXS ports, the MC3810 comes with a built in ethernet port and I believe it does SIP too... Will this mean that I

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as reported? Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke: Peer

Hey, I just started trying to use the qualify=yes option on my Cisco 7960 SIP phones. Of the 13 I have, 2 of them seem to loose their registration with asterisk on a regular basis. I see lots of these lines: -- Registered SIP '3030' at 62.74.107.1 port 58825 expires 60 in my console. But I only see them for 2 extensions. Never see them for the other 11. All 13 phones have the exact same

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

HI, I have been pulling my hair out trying to get a Cisco MC3810 to interface my Asterisk box with a T1. I am able to make outgoing calls but incoing calls never reach my Asterisk box. The cisco give a fast busy when I try to call one of the DID's. When playing around with the dial-peers I can get the cisco to pick up the call, but then it forwards the call back to the ANI that is dialing.

Has anyone here successfully gotten a cisco MC3810 talking with asterisk? I am getting the dreaded - Got SIP response 400 "Bad Request - 'Malformed/Missing URL'" back from xxx.xxx.xxx.xxx If you've gotten it to work you can feel free to email me off list. If your willing to share config's that also is a definate plus. Thanks, --Tammy

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Hi guys, I'm having a really strange problem, which I'm pretty sure has only appeared since my last upgrade (1.2.12.1) . It's about NAT and Qualify. I'm using Asterisk to register with some external SIP providers. However, they're always marked as UNREACHABLE, when they weren't before! A typical debug looks like this: hera*CLI> sip reload Reloading

Hi, I have been trying to get T.38 to work with clients behind NAT for the past week but with no success. I have an asterisk server on the public internet and several Grandstream (I tried Linksys too) HT502 ATAs behind NAT in different locations. I tried every possible combination of NAT, canreinvite, t38pt_usertpsource entries, I even tried asterisk 1.4.19, 1.4.24.1, 1.4.25 all with the same

Hi Everyone, I've got a Cisco 3810 rig with 6 analog FXS ports, and ethernet. I'm wondering in anyone has got one of these suckers to work with asterisk in such a way that each FXS port has it's own extension. It speaks SIP, and I can send calls from asterisk out to it, but can't figure out how to get it to pass username & pw to asterisk when I try to configure it as a

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

Hi All, Anyone have any idea what could cause incoming calls on a SIP channel to no longer be able to use DTMF? DTMF on incoming calls on zaptel and on local SIP softphones and ATAs all work fine. Nothing gets registered in the CDR or on the console in verbose level 10, it just times out. I haven't changed anything on my part and can't get through to Viatalk tech support to ask them

I'm using 1.8.3, and have 2 sip providers. Both are set with qualify=yes. Each of them sometimes have qualify times 10+ times the other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the lower qualify time? sean

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Say, If bandwidth e.g. ADSL goes fuzzy, is there a way to force * to unregister the Peers? I noticed with qualify=200 for example, even if latency goes above and * shows Lagged and then UNREACHABLE The peer's calls are still accepted. Is there a way to automatically prevent this? Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm wondering if anyone has benn able to successfully get g726-16 passthrouhg to work? I am wanting to use this codec instead of g729 as I'm running out of DSPs using a high complexity codec on the Ciscos. I would think it would work just as g729 does, which has been working fine for me, but it does not. G726-32 does work great however, but it's like Asterisk doesn't

We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what

[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer 'gerrie' Am I correct that when I turn on rtcachefriends in sip.conf, database-changes in my MySQL-DB will not be reflected untill a reload ?? Am I correct that when I turn off qualify in my realtime