similar to: Calling non-extension numbers issue

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, 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:

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

Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicemail at mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested

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

Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b)

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

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have

We recently started upgrading OpenSSH on our Sol8 systems and we've run into a problem were we can run commands on a remote system since we installed 3.7.1p1. The debug output from sshd is attached below. We use PAM in our environment, and have since 2.9.9p2. I think most of the systems were running 3.4p1 prior installing 3.7.1p1 and they were working - the only thing we replaced was

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 --

Hi, I am setting up a fileserver in an AD domain, using Samba 3.0.21b under Solaris 9. An Openldap server (version 2.3.17) is used fot the idmap backend. Following the Samba HOWTO Collection, chapter 6, I used the statement ldap idmap suffix = ou=idmap,dc=uppsala,dc=se in my smb.conf. This doesn't work. When starting winbindd I get the following in the log: [2006/02/15 10:31:14, 3]

Hi, I have configured dovecot for auth_bind against an LDAP server on the localhost and I'm seeing the following error in my dovecot logs: dovecot: Apr 30 09:03:18 Error: auth(default): LDAP: ldap_result() failed: Can't contact LDAP server dovecot: Apr 30 09:03:27 Error: auth(default): LDAP: ldap_result() failed: Can't contact LDAP server ... and so on ... When I turned on LDAP

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.

I've got multiple satellite office all linked back to the main office via VPN. Each office has their own asterisk server which registers back to the main office's Asterisk server. Each office also has a 1Mb downstream / 384k - 768k upstream connection. The branches are using Speex for their connections back to the main office. The issue I'm having is that there are times that

I cannot get incoming calls to sip phones behind a PIX to work, outgoing is fine. Asterisk (Public IP) --> Internet --> PIX (NAT) --> Sip Phones I have tried no fixup protocol sip, I have punched a hole in the Pix allowing anything from the Asterisk box into the network, still no incoming. I have done all the Wiki suggests in regarding to NAT. Is their a trick getting the

Can anyone point me in a good direction for configuring SIP through a PIX using 1:1 NAT. I have read anything I could get my hands on and tried them all with very little success. I can get it to work through the cheap little cable modem routers, but not this PIX. I -can- make a direct SIP call using the IP address of the * server (ie.exten@ipaddr), but when I do that * still doesn't

Consider the following directory structure /foo/aaa/*/* /foo/bbb/*/* /foo/ccc/*/* I want to sync all of /foo, but exclude /foo/aaa but not exclude any other occurances of "aaa" or "foo/aaa" (be they files or dirs) that might occur within the other dirs /foo/bbb/* /foo/ccc/* etc I don't want to exclude /foo/bbb/aaa or /foo/ccc/111/aaa or /foo/ccc/111/foo/aaa etc...

Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not

does anyone have a sip working through a cisco pix firewall? i can get the phone to register and the call to be negotiated, but as soon as the call is answered there is no sound and the call ends immediately. im sure this is due to the RTP negotiation being rejected by the pix. any helpful ways around this? right now my only solution is to put a small box outside the firewall and IAX the