similar to: RE: Asterisk-Users Digest, Vol 19, Issue 47

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

I have 2 sip accounts setup - 200 and 202. If I do sip show peers I get: sip show peers Name/username Host Dyn Nat ACL Mask Port Status 202/202 192.168.0.6 D 255.255.255.255 5060 Unmonitored 201/201 (Unspecified) D 255.255.255.255 5060 Unmonitored 200/200 192.168.0.3 D 255.255.255.255 5060

Hi, I've asked this question in the past, but I didn't get a precise answer. Hopefully somebody will take note of my question. Before I forget, I am using Asterisk 1.2.4. I've been using the Voicemail app with success (i.e. it works) except for one single thing: the ONLY message that it ever played back to the caller is the temporary message. If I delete the temporary message

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

Hi the list, Some rumour (!) say that is it possible to prepare some tests for checking our code using .Rin and .Rout. It seems to be a very good practice, but I did not manage to find information on it. So does someone know how it works ? What are we suppose to write in Rin ? More precisely : - I have a package myPack.r in directories ~/myR/myPack/R/ - I create the directory

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

That seems to be the indication given by this webpage http://www.zip.com.au/~akpm/linux/ext3/ext3-usage.html However, default install of Redhat 7.2 setsup fs_passno(6th field of /etc/fstab) as 2 which asks you if you want to run fsck after an unclean shutdown The question is, is fsck required after an unclean shutdown or should one just rely on journal replay. What does fsck do when it sees an

Hello, We have several clients with GXP2000 in their network and behind NAT. We have one particular client that has several GXP2000 behind a Linksys RV082 VPN Firewall/Router which is doing NAT services. According to SIP packet inspection, it detects it's a symmetric NAT. The problem we have is that even though we have configured Asterisk AND the GXP2000 to register every 60 minutes, they

Hi, I have several GXP2000 phones which used to work fine with Asterisk 1.2. However, after upgrading to Asterisk 1.4.21.2, whenever I initiate a call from a GXP2000, it gets dropped after 20 seconds exactly. I have "early dial" enabled on the GXP2000 and "pedantic=yes" on the server. If I disable "early dial", all works well ("early dial" or "overlap

Dear R community, I am using the R package `interval` in order to perform some modelling tests of the NPMLE convergence in the case of censoring. So all I am doing is drawing a sample from exponential distribution, making it a censored sample and computing the NPMLE of its distribution function. But when run on Linux Calculate 10.4 the program keeps crashing and reporting a segmentation fault

Hi, I'm using a GXP2000 (that's sharing the same GXP2020 firmware file) with the latest 1.1.5.10 beta release. It's working since a week and seems working very well. Before I was using the 1.1.5.3 and I had no problem. 1.1.4.xx versions, instead, are not performing like that one (audio, deadlocks and other minor issues). You can find a lot of info and old firmware versions at this

Yes I did. Fair question. I think it`s working, but is there anyway to know for sure? Show modules show app_cdr.so as existing... Mike On Thursday 26 Jan 2006 16:50, Micha?l Gaudette wrote: > Hi, > > I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've > noticed that the CDR logging in MySQL (on a different computer) has > stopped. I thought it

Support obtaining guest storage over ssh. The following now works: virt-v2v -f v2v/virt-v2v.conf -ic 'xen+ssh://xen.example.com/system' \ -op transfer rhel54pv64 --- MANIFEST | 1 + lib/Sys/VirtV2V/Connection.pm | 1 + lib/Sys/VirtV2V/Connection/LibVirt.pm | 4 + lib/Sys/VirtV2V/Transfer/SSH.pm | 208

Hi - The problem is with two abline(s). Attached are: 1) jnk.r to run program 2) jnk.rin the data 3) jnkps.eps the postscript output 4) jnkscreen.bmp (from photoshop after bmp copy to clipboard) in jnk.zip the eps and bmp are different on my machine (windows 2000) any suggestions appreciated bob >>> Diego.Kuonen at epfl.ch 03/18/00 08:24AM >>> "Robert L.

Hi, I use Echo Canceller from speex 1.2rc1. It works excellent, except one kind of test: if I use overloaded (and clipped) RIN and not interlaced local speech and echo in SIN, echo is not removed. I checked the sources (RIN and SIN) with EC diagnostic tool - delay and drift are ok. I would like to understand what the problem is and how to solve it. -- Victoria -------------- next part

1. We just dial the extension directly and have speed dials setup for the first 6 parked positions. We don't use *8 at all. 2. Change the config on the phones under Account to "Send DTMF via RTP (RFC2833)" -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Noc Phibee Sent: Thursday, February 08,

Hi I have One Problem For Example if You Take Category has_many :products Here For Category i have soap as example for products i am taking Santoor , Lux and Rin soaps as products. Here Again Both Lux and santoor comes under Bathing soaps. Rin Soaps Comes under Washing Soaps So these are sub categories... Tom-arrow some new sub category may come like baby soaps.. I am confusing How many