similar to: New RFC: How to specify a phone number

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

hi, i have the problem that i'm not able to set and receive the Service Indication (SIN) from our E1-PRI and from our ericsson BP250. The problem is, that the Bearer Capability (BC) together with the High Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the Service Indicator (SIN). The SIN is used to determine if the call is voice, fax or data. It's essential to set

Where can I get E1 and/or Euro-ISDN specifications/data sheets? Are there specs for other E./G./Q./etc. protocols as well? Thanks!

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Another dumb question... My asterisk system seems to have problems detecting hangups. I am getting a LOT of voicemails with dialtone or silence. I see over at asteriskguru.com there is an explanation of how to configure for polarity reversal in zapata.conf? Does zapata.conf have any function in systems that aren't using zaptel( I suppose not)? I am using an external gateway (wellgate

Note that this attribute is optional. Thanks: Arik Hadas --- v2v/create_ovf.ml | 11 ++++++++--- v2v/test-v2v-o-rhv.ovf.expected | 2 +- v2v/test-v2v-o-rhv.sh | 1 + v2v/test-v2v-o-vdsm-options.ovf.expected | 2 +- v2v/test-v2v-o-vdsm-options.sh | 1 + 5 files changed, 12 insertions(+), 5 deletions(-) diff --git

The information whether the disk is gzip compressed or not is stored in the OVF. There is no reason to do the detection. Signed-off-by: Tomáš Golembiovský <tgolembi@redhat.com> --- v2v/input_ova.ml | 28 +++++++++++++++++----------- v2v/test-v2v-i-ova-gz.ovf | 2 +- 2 files changed, 18 insertions(+), 12 deletions(-) diff --git a/v2v/input_ova.ml b/v2v/input_ova.ml index

The information whether the disk is gzip compressed or not is stored in the OVF. There is no reason to do the detection. Signed-off-by: Tomáš Golembiovský <tgolembi@redhat.com> --- v2v/input_ova.ml | 36 ++++++++++++++++++++---------------- v2v/test-v2v-i-ova-gz.ovf | 2 +- 2 files changed, 21 insertions(+), 17 deletions(-) diff --git a/v2v/input_ova.ml b/v2v/input_ova.ml index

I have a T1 link from asterisk 1.2.23 (also tried with 1.4.13) to a Meridian Option 61C. Calls either way drop with error "Channel 0/23, span 1 got hangup, cause 100". Can anyone offer insight into the cause and solution/workaround? (I tried upgrading to Ast 1.4.13, and upgrading matching zaptel & libpri, put the problem is identical). For testing, I tried a call from the

Hello, I just received a couple SX440isdn phones and was wondering if they can be plugged into a Diva 4BRI port in NT mode and work with asterisk+chan_capi? I know they probably work fine with mutliHFC cards with either bristuff of chan_misdn but since I have some spare Divas, I was curious about their potential as phone ports. The Diva's 3 and 4 ports are already set to NT mode at boot

Hi, On Friday, 24 April 2020 14:57:38 CEST Andrew Thurber (anthurbe) wrote: > This multi-disk ovf generates “no href in ovf:File (id=)” > Other single-disk ovfs on the same system work. I don’t have another multi-disk ova to try. > I’ve compared the syntax with the test file on github and it appears to be essentially the same: > virt-v2v/tests/test-v2v-i-ova-two-disks.ovf > Any

Dear List, I have been trying to extract associations from a matrix whereby individual locations are within a certain distance threshold from one another. I have been able to extract those individuals where there is 'no interaction' (i.e. where these individuals are not within a specified distance threshold from another individual) and give these individuals a unique Group ID containing

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Check the generated OVF for -o rhv and -o vdsm outputs. Variable UUIDs and date/times are filtered out. Make sure the the important UUIDs (disk, volume, VM) are where we think they should be. Signed-off-by: Tomáš Golembiovský <tgolembi@redhat.com> --- v2v/test-v2v-o-rhv.ovf.expected | 92 ++++++++++++++++++++++++++++++++ v2v/test-v2v-o-rhv.sh | 20 +++++++

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

On 7/16/15, Martin Leese <martin.leese at stanfordalumni.org> wrote: > Martijn van Beurden wrote: >> I would propose: 0000-0111 : (number of independent channels)-1. >> The channel order is defined through the >> WAVEFORMATEXTENSIBLE_CHANNEL_MASK vorbis comment, if defined. If >> no WAVEFORMATEXTENSIBLE_CHANNEL_MASK is present, the channel >> order follows

We've got a EuroISDN (32 channels) with a TE405p, running cvs head as of 5 days ago. In the past couple of days, we've hit a scenario where incoming calls to the * pbx from the PSTN are being marked as busy, but outgoing calls work just fine. When we reboot *, the problem goes away. Has anyone else had this ? I've attached a PRI debug below. I've changed the phone numbers (x

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to