similar to: Trouble using 2 IAX2 DiDs provided by different ITSPs

Is there some concensus on where to set callerid when making outgoing calls via an ITSP over IAX? Is this best accomplished in IAX.CONF or EXTENSIONS.CONF? Also, tech support at one ITSP told me that the SetCIDName command doesn't do anything. Is this something that might be unique to their server? Or a general statement of fact? Thanks, Michael -- Michael Graves

Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes

Hi, I am trying to connect two Asterisk servers using IAX2. Everything works fine when I couple them within a LAN segment, but not when I connect them using WAN connections. I made sure that the routers' ports are mapped properly and checked this with additional ssh rules. ServerA is a Raspberry box with the vendor's Asterisk version 1.8.13.1 and ServerB is normal CentOS 7 box with

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up

Hi All, When I want use Asterisk as a PBX to cooperate SIP ITSP, I can not set the caller ID, so SIP ITSP do not accept the call. In Asterisk, I set a account in sip.conf to register on ITSP SIP Server: register => 6292@218.1.121.237/6292 And I added a user 6292 in Asterisk just like the account on ITSP SIP Server: [6291] type=friend username=6291 callerid=6291 host=dynamic

Hello all, Here is my problem, I try to place a call to FWD (free world dialup) trough my asterisk PBX. my config is as follow: extensions.conf ---------------- [internal] exten => 613,1,Dial(IAX2/iaxfwd-outbound/613) (service echo de FWD) exten => xxxxxx,1,Dial(IAX2/iaxfwd-outbound/xxxxxx) mon numero FWD exten => yyyyyy,1,Dial(IAX2/iaxfwd-outbound/yyyyyy) celui d'un ami FWD

With the recent service outage at Broadvoice, there has been a lot of discussion here, on broadband reports, Voxilla, etc., regarding whether VOIP is mature, or "ready for the masses", etc. One particular point I've seen repeated, and with which I agree: "we're willing to deal with less than five 9s, even one or 2 9s, as long as we have good communication regarding the

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

We are using using what Cisco's Port Address Translation, so that all SIP traffic is done through %EXTERNIP%. ?To any outside box, it should look like the asterisk server is actually on %EXTERNIP%. My SIP packet gets sent to the ITSP with a Call-ID: 2fd557964ca936b66661d72f1328c918@%EXTERNIP% , but the SIP 200 OK reply from ITSP has Call-ID: 2fd557964ca936b66661d72f1328c918@%INTERNIP%. ?I can

Has anyone found an ITSP that will relay T.38 fax to an asterisk 1.6.x instance AND do it reliably? If so, I can think of a number of locations with copper loops that could be scrapped. I'm actually quite surprised at what an underwhelming number of ITSP's that say they support T.38 (zero so far among my normal go-to companies). For locations that just want to be able to send

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

Hello list, New to asterisk and to the list (although experienced in Unix/Linux administration). Short problem description: -------------------------- I cannot get the Echo() application to run on any 32bit platform I can get my hands on. In contrast, the only 64-bit (amd64 aka x86_64) setup that I have runs just fine. In all cases asterisk log shows the same -- that Echo() is executed Details:

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

I have an ITSP we are trying to work with that has an "Unusual" way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: <sip:s at x.x.x.x> This then causes ALL

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

Hey all, I was wondering if anyone knows about a Mexican ITSP I can connect to to route calls from and to my * boxen. If it matters: I'm located in The Netherlands and one of our customers is in Mexico so if we need a Mexican presence that is not an issue. Thanks. -- Michiel van Baak michiel at vanbaak.eu http://michiel.vanbaak.eu GnuPG key: