similar to: SIP Registration failure

Displaying 20 results from an estimated 700 matches similar to: "SIP Registration failure"

2009 Mar 18
0
[Fwd: Re: DAHDI or Zaptel doesn't compile against 1.4.24]
Tzafrir Cohen a ?crit : > On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote: > >> Hi, >> >> We installed the latest 1.4.24 on a test machine and can't get zaptel >> nor dahdi compile. It's a Linux Debian Etch. Errors we have: >> >> keewi:/usr/src/dahdi-linux-2.1.0.4# make >> make -C /lib/modules/2.6.18-custom.2/build
2004 Jun 01
2
BroadVoice usage?
Hi all, I've been trying to use BroadVoice as a SIP service provider. They don't officially support * but are helpful when it comes to answering questions for setup parameters. They claim they have no firewalls or access lists that need to be set up so I can get access to their servers. However, something's still not quite right when I use the parameters. It looks like our Asterisk
2006 Feb 13
0
Asterisk register ip phone
Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on
2006 Feb 19
1
Cisco 7960 Register Problem
Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on host it's comming
2005 Mar 08
0
Sip 400 bad request - broadvoice error
I have searched the list and cannot find a sip 400 solution posted that solves my problem. If anyone has any thoughts or suggestions on the following I would greatly appreciate it. I didn't have this error before Broadvoice made their changes this weekend. Now when I make a call it connects but, I cannot hear anything on the other end... The full message I have is: 8 headers, 0 lines
2005 Mar 07
1
working system for months suddenly stopped today with Failed to authenticate on INVITE to - additional
Looks like this is a broadvoice issue... I have been bitten by the changes at broadvoice. I have searched the list and added the three magic lines: >/ username=<phonenumber> />/ authuser=<phonenumber> />/ secret=<registration password>/ But still I cannot place calls out. Calls in are working. Any body have any suggestions... THanks
2006 May 09
1
Asterisk settings Net2Phone
Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer
2006 Nov 13
1
Sending '#' with Dial
Hi! I have a working asterisk-setup with four sip-clients. Everything works great but when the users call someone the phonenumber shows up on the receiving ends callerid-display. To correct this my provider told me to send #31# before the phonenumber, tried this with: Dial(SIP/#31#${EXTEN}@provider) but my asterisk tells me that it isn't a valid extension. The INVITE looks fine,
2005 Sep 18
2
Asterisk Won't Process Call
We have a basic application that runs a SIP channel to pick up a call and process it. We are using Broadvoice and it's been working great. We recently rebooted the machine and now when a call comes in Asterisk picks up the call but does not process it. Asterisk seems to send the call back to Broadvoice. Nothing at all has been changed in the configuration to warrant this. Below is the
2005 Jan 05
0
Asterisk with Euro ISDN, etc
Hi folks! Our company are going to buy an E1 line with Euro ISDN and 30 lines (channels). This is how it will be configured: 3 Lines, of the total of 30, is going to be for the company phones, and share one phonenumber (eg. 555-12340). 1 Line will be dedicated to a specific unique phonenumber (Fax) (eg. 555-54321). The rest of the lines/channels (26) will be used by (by, not for) our customers,
2004 Dec 12
1
I'm stumped
I am trying to use the simple CID name management script on the wiki. http://www.voip-info.org/wiki-Asterisk+tips+managing+CID+names I can not see what is wrong. The values never get entered in the database. Here are the files: I have asterisk running as the user asterisk also. ---cid-store.php---- <HTML> <HEAD> <TITLE>Storing Asterisk CID data</TITLE> </HEAD>
2003 Nov 03
1
one way sound with x-lite (sip) -3rd attempt !
Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation
2006 Jan 17
2
change error messages for Validation helpers?
Is it possible to change error messages for Validation helpers? I am writing an app against a existing database (so no control over column names), but when there is validation error (e.g. with validate_presence_of) I would like to customize the field name. For example for telephone whose field name is PhoneNumber I would like to chnage it to "Telephone Number cannot be empty" rather
2014 Aug 11
1
401 Unathorized
I have an asterisk 1.8.x box that intermittently returns a 401. Calls come through the same peer all the time, from the same carrier. However intermittently the asterisk box returns a 401. Below is the output of a failed call (1st) and a successful call (2nd). I can't see any difference until we get to these lines. Bad call: --- (17 headers 14 lines) --- Sending to carrierIP:5060 (no NAT)
2006 Oct 18
0
[OT] Nokia E60/61/70 and SIP
Martin Joseph wrote: > > > For all of us using these devices, I have some good news. There is a > self installable firmware update available from Nokia here (requires > windows box to install): > > http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate > > This seems to radically improve the behavior of the SIP client on my > E60. It seems to have
2008 Jan 18
1
Automatic call-out problem
Hello! My setup is Asterisk 1.2.26 with Zaptel 1.2.22.1, libpri-1.2.7 on Fedora Core 4. I am making automatic call-out campaign with this setup on 4 PRI. The scripts for this: ==================================================================== caller php script write this to outgoung folder: fwrite($outfile,"Channel: Zap/g1/$phonenumber\n"); fwrite($outfile,"MaxRetries:
2009 Sep 21
0
Polymorphic form
I have two partials to deal with contact information. The Contact information uses single-table polymorphism. I want to be able to use save on Contact and set ''type'' manually, based on the type of form the user is filling in. This is saved as the value on a hidden field. class Contact belongs_to :person acts_as_list :scope => :person validates_presence_of :type
2006 Jun 26
2
n-way has_mant :through
I''m trying to setup some mildly complex associations for a project we''re working on and can''t seem to find much documentation on n-way has_many :through associations. I have the following models: Person, PhysicalAddress, EmailAddress, PhoneNumber. Each person can have multiple PhysicalAddresses, EmailAddresses, and PhoneNumbers, and multiple people can share the same
2005 Feb 01
2
Outbound calling with TDM400P
I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",
2006 Feb 02
0
Sip - no peer or user found on incoming call
Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I