similar to: iax2 trunking issues

Paul, Thanks - it worked! For the record, this is exactly what I did: cd/usr/src/asterisk/cdr grep -in "userfield" cdr_csv.c (to find the line that had "#define CSV_LOGUSERFIELD 1" commented out) Opened cdr_csv.c and removed the /* and */ comment marks Saved & exited After shutting down * I went to /usr/src/asterisk and did the usual: Make clean Make Make install I also

Hi I am trying to get a SNOM 360 to monitor other extensions i.e. when someone makes a call to/from another extension, one of the LED's on the SNOM 360 will change state. I am using 1.0.9/bristuff-8l. I have 2 extensions - 2001 is a SNOM 190, 2002 is a 360 - both are running the latest firmware. (3.60i for 190, 4.0 for 360). I have read all the relevant articles on the wiki on

Hello I see it is possible to buy Flash Disks up to 4GB now. Has anyone any experience of building an Asterisk system with a flash disk as the only storage device? Any brands you recommend? Is 2 or 4GB enough for an Asterisk installation? Typically how many MB is required for voicemail recording files for say a 10 user system? What about voicemail - I suppose files could be emailed and

I would like to know if somebody else experienced that: sip show peers will always drop the first character of the 11th line. while sip show peers like [0-9,a-z] will not drop any character. Can anybody test this, please? bye Ronald Wiplinger

hi all, I have a problem with @ 1.2.4 on debian kernel 2.6.8-2-386.: -- Executing Dial("SIP/2003-bbae", "zap/2/03460816149|30|t") in new stack Feb 14 17:25:25 WARNING[4246]: channel.c:2535 ast_request: No channel type registered for 'zap' Feb 14 17:25:25 NOTICE[4246]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'zap' (cause 66 - Channel

I have A@H 2.1, running * 1.2.1. I am trying to put information into the userfield with SetCDRUserField and AppendCDRUserField. However, the field is never populated in the cdr - I've checked the csv files and the MySQL asteriskcdrdb table. The field is defined in the MySQL table, but is always empty. The csv files that get created don't have a userfield at all, that is, there

http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz.sig v1.0.8 and v1.0.9 were a bit bad releases. Hopefully one day I've managed to have written a proper test suite which can be run before doing any releases.. * Security hole with LDAP+auth cache: If base setting contained %variables they weren't included in auth cache key,

http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz http://dovecot.org/releases/1.0/dovecot-1.0.10.tar.gz.sig v1.0.8 and v1.0.9 were a bit bad releases. Hopefully one day I've managed to have written a proper test suite which can be run before doing any releases.. * Security hole with LDAP+auth cache: If base setting contained %variables they weren't included in auth cache key,

Lilantha, the tones are supposed to be switched using the loadzone and defaultzone lines in /etc/zaptel.conf , and, progzone in /etc/asterisk/zapata.conf. The information about countries and frequencies/times are at zonedata.c located in the sourcecode of zaptel. As you may know, changing zonedata.c information requires a re-compilation of the zaptel module. Hope it helps, Ricardo

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if

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

Hello! I have bin trying to set up trunking between some of my Asterisk boxes but had no luck... I use Asterisk 2.0.6 on SuSE 6.2, but I have had the same problem with erlier releases. I have a working connection and can place multiple calls in both direktions. Than I set "trunk=yes" on both sides and reload. "CLI> iax2 show peers" shows a (T) and low latency on both

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-

IAX2 now has support for a "trunk" mode ("trunk=yes" in the appropriate friend section). Trunk mode allows IAX2 to use bandwidth extremely effectively. The original impetice (and strategy) was a result of a mistake in which it was claimed that Asterisk with a T100P could send 96 simultaneous calls over a single T1 using VoIP. Thanks to a suggestion by the customer, combined

Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___________________________________________________________________________

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, Just hit this problem for the first time: WARNING[17712] chan_iax2.c: Too much delay in IAX2 calltoken timestamp from address 10.25x.xxx.160 When I ran "iax2 show peers" everything comes up as unreachable, no calls are passed between the servers (as would be expected) but there is no problem with DAHDI / SIP channels. At the same time there are no network issues (can ping all