similar to: Call forward & SER as SIP router

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

I just setup CENTOS 4.7 with latest patches on DELL server. I also configured NTP point to out time server. I found /var/log/messages file every 20 to 30 minutes will generate a error message : Mar 15 14:28:15 SER1 ntpd[25037]: sendto(172.29.21.16): Invalid argument Mar 15 14:45:22 SER1 ntpd[25037]: sendto(172.29.21.16): Invalid argument Mar 15 15:02:29 SER1 ntpd[25037]: sendto(172.29.21.16):

Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone

Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: >Pamela, >Did you resolve the problems you described? >I didn't see a reply on the list but I may have missed it. > >-Kevin > >-----Original Message----- >From: Pamela Weis [mailto:peawy@gmx.at] >Sent: Thursday, August 05, 2004

Hi, I've been registering asterisk to ser. I'm using SER as the outbound SIP trunk for Asterisk. Users registered with Asterisk will use the SIP trunk to reach SER registered users and PSTN's. Now when I register Asterisk with SER, on my SER's location table I see these record: Username Column = asterisk Contact Column = sip:s@202.84.24.47 I have a script running that checks

Hi all, I have a time series that contains double seasonal components (48 and 336) and I would like to decompose the series into the following time series components (trend, seasonal component 1, seasonal component 2 and irregular component). As far as I know, the STL procedure for decomposing a series in R only allows one seasonal component, so I have tried decomposing the series twice. First,

Using Asterisk 1.0.9 When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten

> Hello fellow asterisk people! > > I have Asterisk listening on port 5061 and SER on port 5060. > > Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP. > > My problems are with SIP. I can make incoming calls from SIP to asterisk > and to any of the other networks, but when I try to make an outgoing call > from Asterisk to SER I see the following in

Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce,

Hi All, I have CentOS 4.2 with ser 0.9.6 and asterisk 1.2.4. Ser is listening on 5060 and asterisk on 5065. The setup is that people use serweb to create an account and register a phone. Their calls are routed from ser to asterisk and then inbound on IAX2. The server has a public and an internal interface. The real FQDN of the server is nmibwksip3.nexusmgmt.com and it has cnames of pbx and

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: 13(2894) DEBUG: RFC3261 transaction matching failed 13(2894) DEBUG: t_lookup_request: no transaction found 13(2894) SER: forwarding ACK

I have some problem to redirect the call from asterisk to ser. 1 thing i am redirecting call to asterisk and then on some extension i want to return the call to ser. Receiving this error: WARNING[23594]: chan_sip.c:6829 handle_response: Forbidden - wrong password on authentication for INVITE to '"Alex" <sip:xxxxxxx@xxx.xxx.xxx.xxx:5061>;tag=as55a3adbb' --

Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable "call forward". The result of CDR seems not correct. UA 1011 make a call to UA 9999, and UA 9999 forwards this call to a PSTN number. I think we shall charge the credit from UA 9999 not UA 1011 because UA 1011 don't know where UA 9999 forwards to. But in CDR, I can only find the from(1011) and

Well here's a suggestion - a little crazy - but works... Most equipment is taking the 120vac and converting it into DC voltage. So why not just feed it DC voltage directly??? We had a situation where our field techs needed to test dsl circuits and voip ata from the demarcation point outside a house or business. A UPS might have worked - but the down conversion of 12v dc battery in ups up to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Hello, I have asterisk connected to SER/RTPProxy which is again connected to a IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone connected to the IP-PSTN gateway, I get 'empty ACKs': U 192.168.0.173:5060 -> 10.254.254.1:5060 ACK SIP/2.0. Via: SIP/2.0/UDP 192.168.0.173:5060;branch=z9hG4bK5cb7d048. Route:

Hello, here my achitecture: client1--Asterisk1----ser1---centile client2-- client1 do a call to centile. centile do a forward to client2 (Diversion) and then use the same CALL-ID! when asterisk1 receive the call with the same CALL-ID, it screen "Now forwarding SIP/XXXX -000002f6 to 'Local/MCDU at kamailio ' (thanks to SIP/YYY-000002f7)" I don't want that asterisk receive