Carlos Chavez
2006-Apr-18 15:24 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
I am having a strange problem with A@H 2.7 (Asterisk 1.2.5) with a TE210P card and Unicall. I have compiled everything and Unicall seems to be working well. The only problem we are having is that the outgoing voice is a bit distorted. When someone from the inside calls (all phones are connected to ATA using SIP) they can hear the other person perfectly, but the remote hears them distorted. I have checked and rechecked the configurations and everything seems fine. I really do not know if this is a software problem, a hardware problem or a line quality problem. Anyone has any idea how to pinpoint the source of the problem? -- Carlos Chavez Prats Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 191 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060418/1a364c50/attachment.pgp
Moises Silva
2006-Apr-18 17:15 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
Hi Carlos. Please provide the version of unicall. Also, are you using MFCR2 or PRI? and finally, provide us with the codecs that you have tried. PSTN in Mexico should be ALAW, so try to avoid transcoding and use ALAW in the phones too, we had bad issues with ILBC when the calls were coming from Avantel until we started using ALAW. Regards On 4/18/06, Carlos Chavez <cursor@telecomabmex.com> wrote:> I am having a strange problem with A@H 2.7 (Asterisk 1.2.5) with a > TE210P card and Unicall. I have compiled everything and Unicall seems > to be working well. The only problem we are having is that the outgoing > voice is a bit distorted. When someone from the inside calls (all > phones are connected to ATA using SIP) they can hear the other person > perfectly, but the remote hears them distorted. I have checked and > rechecked the configurations and everything seems fine. I really do not > know if this is a software problem, a hardware problem or a line quality > problem. Anyone has any idea how to pinpoint the source of the problem? > > -- > Carlos Chavez Prats > Director de Tecnolog?a > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.2.2 (GNU/Linux) > > iD8DBQBERWcDVhw7eWImqUMRAraUAJ0SZ6PBbKLfafv5UJ9/+QTJ911IrwCfcZWu > oLNvahpE8FvWoMjfGkQvE10> =Slpo > -----END PGP SIGNATURE----- > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
Carlos Chavez
2006-Apr-18 19:13 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
On Tue, 18 Apr 2006 19:15:33 -0500, Moises Silva wrote> Hi Carlos. Please provide the version of unicall. Also, are you using > MFCR2 or PRI? and finally, provide us with the codecs that you have > tried. PSTN in Mexico should be ALAW, so try to avoid transcoding and > use ALAW in the phones too, we had bad issues with ILBC when the > calls were coming from Avantel until we started using ALAW. >We are using SpanDSP 0.0.2pre25 and unicall-0.0.3pre9 (libsupertone, libunicall and libmfcr2). I have tried several versions of each library but those are the same ones I have on other servers that work perfectly. All Linksys PAP2 units use ALAW as their primary codec. The server is an HP ML series with 2 processors. Spans are as follows: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" span=1,1,0,cas,hdb3 cas=1-15:1101 dchan=16 cas=17-31:1101 # Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 2" span=2,2,0,cas,hdb3 cas=32-46:1101 dchan=47 cas=48-62:1101 I have tried both E1 ports and I get the same distortion. Here is my unicall.conf: protocolvariant=mx,10,4 protocolend=cpe usecallerid=yes hidecallerid=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default context=from-trunk channel => 32-46 ;skip time slot 47 channel => 48-52 There are only 20 channels available at the moment. To eliminate the phone company as the source of the problem I connected both ports using a T1 crossover cable. Here the problem got worse as now both sides of the conversation are distorted. When on a normal call to an outside phone only the sound from the internal phone is distorted. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001
Stepan Hradsky
2006-Apr-18 23:37 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
Hi, I had similar problem and problem was in SIP ATA device (we use Sipura 2100). They was set from factory to send 30ms voice frame, when we change frame to 20ms everything work perfectly. Stepan Carlos Chavez napsal(a):> I am having a strange problem with A@H 2.7 (Asterisk 1.2.5) with a > TE210P card and Unicall. I have compiled everything and Unicall seems > to be working well. The only problem we are having is that the outgoing > voice is a bit distorted. When someone from the inside calls (all > phones are connected to ATA using SIP) they can hear the other person > perfectly, but the remote hears them distorted. I have checked and > rechecked the configurations and everything seems fine. I really do not > know if this is a software problem, a hardware problem or a line quality > problem. Anyone has any idea how to pinpoint the source of the problem? > > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Oddeleni spravy site a pece o zakazniky ha-vel internet s.r.o. internet x voice x family x cafe . rEvoluce Svabinskeho 9 702 00 Moravsk? Ostrava tel./fax: +420 552 305 306 email: stepan.hradsky@ha-vel.cz www: http://www.ha-vel.cz Oddeleni pece o zakazniky: +420 552 305 345 Dohledove centrum: +420 552 305 321 Neodstranujte prosim zadnou cast tohoto e-mailu pri pripadne dalsi komunikaci k tomuto tematu. Please do not remove any parts of this e-mail message in further communication about this issue.
If the voice distortion sounds like "clack clack clack" las if you had a fan right next to you (remember when you talk directly to a fan in front of you, the other side gets your voice like in intervals), if that?s the case, exactly, your frame size should be 20ms, sipura and some other atas come by default with 30 packet sizes, after changing to 20, all worked fine. Hope that helps. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Stepan Hradsky |Sent: Wednesday, April 19, 2006 1:38 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Outgoing voice distortion with Unicall | |Hi, | |I had similar problem and problem was in SIP ATA device (we |use Sipura 2100). They was set from factory to send 30ms voice |frame, when we change frame to 20ms everything work perfectly. | |Stepan | |Carlos Chavez napsal(a): |> I am having a strange problem with A@H 2.7 (Asterisk |1.2.5) with a |> TE210P card and Unicall. I have compiled everything and |Unicall seems |> to be working well. The only problem we are having is that the |> outgoing voice is a bit distorted. When someone from the |inside calls |> (all phones are connected to ATA using SIP) they can hear the other |> person perfectly, but the remote hears them distorted. I |have checked |> and rechecked the configurations and everything seems fine. |I really |> do not know if this is a software problem, a hardware problem or a |> line quality problem. Anyone has any idea how to pinpoint |the source of the problem? |> |> |> |---------------------------------------------------------------------- |> -- |> |> _______________________________________________ |> --Bandwidth and Colocation provided by Easynews.com -- |> |> Asterisk-Users mailing list |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users |> | |-- |Oddeleni spravy site a pece o zakazniky |ha-vel internet s.r.o. |internet x voice x family x cafe . rEvoluce | |Svabinskeho 9 |702 00 Moravsk? Ostrava | |tel./fax: +420 552 305 306 | |email: stepan.hradsky@ha-vel.cz |www: http://www.ha-vel.cz | |Oddeleni pece o zakazniky: +420 552 305 345 Dohledove centrum: |+420 552 305 321 | |Neodstranujte prosim zadnou cast tohoto e-mailu pri pripadne |dalsi komunikaci k tomuto tematu. |Please do not remove any parts of this e-mail message in |further communication about this issue. | |_______________________________________________ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |
Carlos Chavez
2006-Apr-19 08:30 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote:> Hi, > > I had similar problem and problem was in SIP ATA device (we use Sipura > 2100). They was set from factory to send 30ms voice frame, > when we change frame to 20ms everything work perfectly. >Where in the Sipura configuration is that option? I cannot seem to find it. -- Carlos Chavez Prats Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 191 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/96b7c414/attachment.pgp
Rich Adamson
2006-Apr-19 08:58 UTC
[Asterisk-Users] Outgoing voice distortion with Unicall
Carlos Chavez wrote:> On Wed, 2006-04-19 at 08:37 +0200, Stepan Hradsky wrote: >> Hi, >> >> I had similar problem and problem was in SIP ATA device (we use Sipura >> 2100). They was set from factory to send 30ms voice frame, >> when we change frame to 20ms everything work perfectly. >> > > Where in the Sipura configuration is that option? I cannot seem to > find it.Login as admin and advanced. On the SIP tab, under RTP parameters, the entry "RTP Packet Size:". Default value is .030, change that to .020 (for 20 milliseconds).