similar to: g729 with Cisco gateways?

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question.

Hello All, I looked at the codec supplied by Digium and I think that it is G.729 Annex-B (G.729b - 8kbps). Correct me if I'm wrong. I have also looked at the article from Cisco "VOIP Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation", which states that G.729b is supported on pretty much everything except for AS5350/5400. Also that G.729b is a "high

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Greetings, After the 0 respones I had on my previous mail regarding the Cisco 2801, I thought I'd be more general. Is anybody using Cisco media gateways at all? If so, how is it working for you? -- Kind regards, Meilleures salutations, Bas van der Veen GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x9E890160 "The question of whether a computer can think is no

I am running an Asterisk server through a Cisco 7206 PSTN gateway. I am able to make outgoing SIP calls without a problem, though incoming calls have been somewhat of a problem. I am not sure exactly how sip.conf should look in such a scenario. I believe most Cisco gateways are just managed through ACL's, with no authentication, so I think I have the outgoing "peer" statement

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

Kai, Thank you for the reply. I didn't want to bother the list too much. However, after reading I discover I don?t have a clear cut way of doing transcoding. Can somebody direct me to where I can get document to get this transcoding done. My set up >From [cisco (g729)] ----> [asterisk (sip channel(g729)within the same asterisk) g711 to chan_ss7] -----> [pstn] And vice versa. I

I'm trying to place a call from my Cisco 7960 and I'm receiving this error: Mar 1 06:19:44 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel Mar 1 06:19:58 NOTICE[3060]: chan_sip.c:7399 handle_request: Unable to create/find channel I can't place calls, but I can receive them: mail*CLI> sip show channels Peer User/ANR Call ID Seq

Hi all, Just wondering if * can work with Cisco Gateways such as Cisco 2600/3600 routers or a VG200? ---------- Edward J. Gomez Director of Network Services ProxyMed, Inc 2555 Davie Road, Suite 110 Fort Lauderdale, Florida 33317 (954) 473-1001 x315 (954) 473-1656 FAX <http://www.proxymed.com/> Confidential, unpublished property of ProxyMed, Inc. (c) copyright as of the date

More top posting goodness... Please post your updated dialplan. After making the change, did you reload/restart Asterisk so the changes would take effect? --Tim ----- "Mohit Saxena" <MohitS at starcomms.com> wrote: > Still no luck.... > > Br, > Mohit > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > Possibly slightly off topic, has anyone ever had Cisco 79xx Series > phones come up with ?cannot

All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's & ATA's) via G729 -->Cisco Gateway-->Asterisk via G711. Any ideas? Has anybody done such an implementation or know

We're running Asterisk 1.2 on a network with a Cisco Voice Gateway (2811) and 7940/7960 SIP Phones, and were wondering what the recommended software versions on these were. Currently, we have 12(4)2T4 on the Voice Gateway, and 7.5 on most of the phones. Is this a fairly standard, supportable setup, or should we look for an upgrade? -------------- next part -------------- An HTML attachment

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

I have several Cisco SIP gateways sending calls to Asterisk. Because the gateways don't have user-agents, they don't authenticate with Asterisk. And because they don't authenticate, they use the default context in the sip.conf file. Is there a way to either: A) identify the inbound gateway with a variable, in channel info, or the manager interface? If there was a ${SIPDOMAIN} for

I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@10.1.1.1,240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Hi all! Quite a mystery. The following happened when I was on holiday, and no one else has changed any configs of either Asterisk or the Cisco's in the building... The situation: Incoming works fine on all phones. Outgoing only works from non-Cisco phones. When calling from a Cisco phone to an external phone, all the Cisco-user hears is a ticking crackle and after about a minute the phone

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form