similar to: TE4XXP: Version Synchronization Error!

Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [root at catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/ RED 1 PRI CAS RED 2 PRI CAS RED 3 PRI CAS RED 4 PRI

Hi, would be glad to be contributing with this question to all the comunity. I?m having a weird issue, suddenly I get channels on IDLE 0x00 state until I do a dahdi restart. Trying to do a call trace to see whats going on deeper, I get surprised when tried to open the .call file to see the log this is incomplete. This is what I see [root at localhost telefonica]# nano

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet ->DSL transparent bridge ->router ->asterisk ->softphone x-lite attempts to login and register, but times out. There must be some setting I'm

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

I wonder if Asterisk's skinny/sccp channel driver could be used as a "client" to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hello I have a simple configuration to allow the admins to listen the agents calls: exten => _654,1,ChanSpy(Agent) exten => _654,2,Hangup() The problem is... even when the agents hung up... it seems the channels remain active: asterisk*CLI> show channels SIP/211-b3042018 654 at default:1 Up ChanSpy(Agent) SIP/211-b3fbf768 654 at default:1 Up ChanSpy(Agent)

Hey gang, I'm trying to find a) If you can put channel variables into a Call file and b) what the appropriate syntax is. Any ideas? Thanks, PB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090721/cb8c2656/attachment.htm

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

I just needed to test a dialup modem connection (don't ask) and I had a modem connected to a TDM card (FXS port) which then dialled out via a E1 PRI on a TE4xxp card. See my log below: atdt0198xxxxxx CONNECT 36000 V42bis ** Dial IP ** Username: xxxx Password: Entering PPP Session. IP address is xxx.xxx.xxx.xxx MTU is 1500. OK ATH0 OK ATDT0198xxxxxx CONNECT 31200 V42bis ** Dial

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page