similar to: chan_ss7 0.10.1

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

Hello everyone, I am trying to apply this<http://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diff>patch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors: Hunk #1 FAILED at 704. Hunk #2 FAILED at 715. I have done the patch modifications manually in l4isup.c There is just one question, how do I pass the RB file-to-play on an SS7 channel via

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server with Digium E1 cards ( 4 port cards). This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total

hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is

Dear all, Do you have come acrross with this issue. My ss7 link get fluctuating. It use chan_ss7 version 1.0.95-beta. I have 8 E1s running on a DL380 server. This enable to have calls from sip to ss7 and vice versa. However ss7 links are not stable. linkset siuc, link l1, schannel 1, sls 0, NOT_ALIGNED, rx: 1, tx: 2/4, sentseq/lastack: 127/127, total 4034145216, 4031118560 linkset siuc, link

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

Dne 28.1.2016 v 13:37 Brian :: napsal(a): > when you say load - how many concurrent calls? Is there transcoding > happening? sip / PRIs ? what load? > 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) > On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > >

hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --------------------------------------- Marek Cervenka =======================================

2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > on my own server > Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent

hi, i''m trying move from file: based disk to tap:aio but things don''t work i have centos4 dom0 with centos4 domU xen 3.0.4-testing changeset: 13138:d401cb96d8a0 self compiled [root@xen linux-2.6.16.38-xen]# grep XEN_BLKDEV_TAP .config CONFIG_XEN_BLKDEV_TAP=m config disk = [ ''file:/var/lib/xen/test.img,hda1,w'',

hello, is there some way to notify people in the same pickup group about call from caller to callee? i.e. i have call from 111 to 222 there are 222,333,444 in the same pickup group 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup the call with *8 siemens have this on their sip openstage phones. how they do this? thanks -- --------------------------------------- Marek

Hi team, Can any one share with me clustering configuration file SS7.conf for single pointcode with four slc. two different machine each host having 2 slc respectively. Thanks Vinod Dharashive Sent from BlackBerry? on Airtel

Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to wav >>>> (centos6) >>>> when the system is

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

Dne 4.2.2016 v 12:17 A J Stiles napsal(a): > On Thursday 04 Feb 2016, Marek ?ervenka wrote: >> hi, >> >> is there way to get SQL schema for Asterisk 13.7.0 without alembic? >> thanks > Assuming you already have Asterisk up and running, you can just use > > $ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ... > > will print (on STDOUT, so you can just

hi, can you recommend "clean&simple&stable" solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --------------------------------------- Marek Cervenka =======================================

This was always possible in the past, however does not work in the current release. I believe this is a bug. To: asterisk-users at lists.digium.com From: cervajs at fpf.slu.cz Date: Fri, 9 Oct 2015 10:04:47 +0200 Subject: Re: [asterisk-users] Storing HANGUPCAUSE in CDR search in archives save the records to another table like cdr_extended Dne

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287