similar to: strange text message:)

hi, I'm been successful in making calls to another local extension using Nokia E71. However calling the E71 from another ext. (X-lite) is not successful. There is a ringing tone from the caller side but the E71 is silent. Tried disabling the NAT (dunno whether that helps). Instructions where from http://www.geek.com/articles/mobile/feature-voip-with-nokia-e71-how-to-2008095/ Am i missing

In SIP setting on the e71 I set the public user name as 1995 at 10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '<sip:%201995 at 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:

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

Greetings list, I have recently acquired an Nokia E71 (which comes with Symbian 3rd edition, feature pack 3 I believe). Accessing my emails has worked before, but now, I cannot connect to the mail server any longer. If I enable verbose_ssl, I get the following error in the log: SSL_accept() failed: error:140943F2:SSL routines:SSL3_READ_BYTES:sslv3 alert unexpected message [141.84.69.67] I

Hi all, Thanks to the previous replies that helped me with this before, but I got side-tracked in the middle of trying to figure this out, so apologies for posting the same issue. I use a Nokia e71, with an asterisk server and am having an issue dialing certain numbers. When I attempt to dial a local number, like xxx-xxx-xxxx, I cannot connect. What shows in the asterisk debug is the

Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

Hello ppl, I want to make a special dial plan for routing calls to a peer which has an pin protection. Normally if you want to call through that peer you must first enter pin for example 1234# and after that you hear the tone from line and after that you can dial desired numbers. I tried something like that, but doesn't worked. Did somebody have some clues? exten =>

I am encountering problem recently with the chan_mobile that the bluetooth connection between the asterisk and my Nokia E71 mobile phone frequently connect and disconnect within seconds. As a result, I can't make any call using Mobile/E71/{exten:2}. Any suggested cause? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an asterisk server at home. I'm looking to replace my internal phones with sip cordless (DECT) phones. I'm now looking at the Siemens A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base ($80) and DP710 handset ($45). The Siemens has a feature were I can also use a PSTN landline, but I not sure that's a great benefit. Has anybody tried these phones? I

Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get

Hello, is anyone on the list using a normal cell/mobile phone which is able to act as a SIP client over WLAN? Or has anyone heard of a SIP client for cell/mobile phones running windows mobile 6.x? The phone should use SIP, when the asterisk server is reachable and should automatically switch to a German telco if it is not reachable. Thanks for any hints, Stefan --

Hello, In Arch Linux. The yajl library distributed is a newer than used by Xen in mercurial. Is there an interest in this patch. I also have a PKGBUILD that builds Xen from mercurial in Arch Linux. Regards, -- Felipe Magno de Almeida Sent from my Nokia E71

Hi everyone, after doing the same thing multiple times and struggling to remember how it was done, I have prepared a small tutorial that explains how to save monitored files in different folders per day. This is quite useful becausethe resultingfile system is way more manageable than having maybe 100,000 files all saved in the same folder. You can find the tutorial here:

Hi guys, Has anyone had any luck with getting the Cisco IP Communicator working with your Asterisk or primarily, Trixbox installation? I've tried searching the net for information, and found someone said to set it up like the 7970 hard phone, which I have, and I'm just running into the problems with it saying "Error Verifying Config Info". Any and all help is appreciated.

Dear all! I have a data frame composed by 13 columns (year, and 12 months). I want to transform this data base in another like this year month values 1901 1 1901 2 1901 3 ..... 1901 12 1902 1 1902 2 .... 1902 12 Is there a possibility to succeed that in R? Thank you! best regards! CR -- --- Catalin-Constantin ROIBU Lecturer PhD, Forestry engineer Forestry Faculty of Suceava Str.

I''m a new comer. I have problems using tc+htb. I run the following commands, and expect outgoing ssh flow goes into 1:11. But actually it goes into default 12. What''s wrong? tc qdisc add dev eth0 root handle 1: htb default 12 tc class add dev eth0 parent 1: classid 1:1 htb rate 1000kbit ceil 2000kbit prio 1 tc class add dev eth0 parent 1:1 classid 1:11 htb rate 100kbit ceil