similar to: Asterisk IAX Trunking Stops - Too much delay in IAX2 calltoken timestamp from address

Hi i have a small problems on two Asterisk Server 1.6.4 : The first sent the call to the second, and in the second, i have a error : [Nov 15 15:30:12] ERROR[5113]: chan_iax2.c:4529 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address IP_FIRST_ASTERISK in the calltokenignore list or setting user 04TELNUMBER requirecalltoken=no on the second

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

I had two asterisk servers connected with each other, both were 1.4.22 but I've upgraded one to 1.4.37 and now I get a message when I try call from asterisk-1.4.22 to asterisk-1.4.37 ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.140.1 in the calltokenoptional list or setting user guest

Hello, I have some problems in placing calls through IAX... It does not work :) in the asterisk console I can't see nothing about dialplan enter or so, IAX debbugging seems to be unuseful... this is my configuration: [612] type=friend secret=123456 notransfer=yes disallow=all allow=gsm allow=ulaw allow=alaw context=from-internal host=dynamic requirecalltoken=no I enabled IAX debugging, but

B.H. Hello! We have several Asterik boxes that are connected to PSTN using PRI cards and they are interconnected using IAX2 trunks so that incoming calls are delivered from PSTN to the servers they belong to. In past we were using asterisk 1.4 on the server that is receiving IAX connections and everything worked as expected. Recently, we have switched to a newer box with asterisk 1.8.22 and

Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I originally wanted to answer with something ... tzarit and kevit Readed probably before you invent rapid biz. I am asking to share any info /experience not your high spirit. Thanks for less trivial answer, G >On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: >> I am using the inter asterisk trunking and the article in >> voip-info.org<http://voip-info.org>will

Hi All; I am configuring IAX endpoint, I just need to understand why I have to set requirecalltoken = no to be able to register because the following message is displayed: [Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 78.154.240.146 in the calltokenignore list or setting user gwbilalkwpciax

I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken

Hopefully this is really a generic question about IAX and doesn't turn out to be something specific to Teliax, because they haven't been too helpful so far. All they can tell me is that my login shows "status unknown" on their end, which prevents me from receiving inbound calls on my Teliax number. Outbound calls through the same server work fine, which rules out most networking

Hi All, Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and asterisk 1.2.14 ? i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123

For those that might be using Cisco 7940/7960 sip phones and placing calls across an iax2 link, we think the voice quality problem has been identified and corrected. The dev cvs should be updated as of about 3:30pm CDT today (April 14). History: Calls originating from a Cisco 79x0 sip phone and sent via iax2 link to some distant * machine resulted in very poor quality audio, and in some cases,

Hi All I am using RHEL , kernel 2.6.9-5, asterisk 1.2.4 , zaptel 1.2.3 installed , when I give modprobe for zaptel and ztdummy , I do not any error message my iax.conf contains the entry for trunking as [hoportal] type=friend host=192.168.20.32 secret=mysecret context=local trunk=yes my extensions.conf contains the entry for trunking as exten =>

Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is "UNKNOW" like: * Name : marco Secret : <Set> Context : phones Parking lot : Mailbox : 2345 at default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk : No Encryption : No

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hi, we have following setup: fax machines > PSTN -> GW SIP -> Asterisk -> Peer IAX (Elastix) Asterisk is 11.16 as well as Asterisk version of Elastix peer. When sending incoming fax calls to the Peer IAX -which receive them using hylafax- we want to tell our asterisk to NOT detect fax CNG. It's easy on a SIP channel as faxdetect=no can be set for the peer, but how to do it

Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Server B can place calls to Server A but when trying to place calls from Server A to Server B this is what I am getting:

I have 11.4.0 on an Amazon EC2 instance. SIP works fine, but I can't get iax to work. I've opened 4569 in the EC2 Security Group. I'm using the zoiper client. Using tcpdump I can see the zoiper packets coming in on 4569, but nothing shows on the asterisk cli. Frame 33: 79 bytes on wire (632 bits), 79 bytes captured (632 bits) on interface 0 0000 12 31 3b 12 40 84 fe ff ff ff

Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -------------- next part -------------- An HTML attachment was

Hi all! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *